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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.
This commit is contained in:
Wim Taymans
Tim-Philipp Müller
parent
2a8cfc6410
commit
0c4fe985b6
7 changed files with 97 additions and 93 deletions
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@ -18,34 +18,34 @@
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*/
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/**
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* SECTION:element-rtpbin
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* SECTION:element-gstrtpbin
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* @short_description: handle media from one RTP bin
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* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
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* @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
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*
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* <refsect2>
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* <para>
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* RTP bin combines the functions of rtpsession, rtpssrcdemux, rtpjitterbuffer
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* and rtpptdemux in one element. It allows for multiple rtpsessions that will
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* RTP bin combines the functions of gstrtpsession, gstrtpssrcdemux, gstrtpjitterbuffer
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* and gstrtpptdemux in one element. It allows for multiple RTP sessions that will
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* be synchronized together using RTCP SR packets.
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* </para>
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* <para>
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* rtpbin is configured with a number of request pads that define the
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* functionality that is activated, similar to the rtpsession element.
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* gstrtpbin is configured with a number of request pads that define the
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* functionality that is activated, similar to the gstrtpsession element.
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* </para>
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* <para>
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* To use rtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
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* To use gstrtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
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* number must be specified in the pad name.
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* Data received on the recv_rtp_sink_%%d pad will be processed in the rtpsession
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* manager and after being validated forwarded on rtpssrcdemuxer element. Each
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* RTP stream is demuxed based on the SSRC and send to a rtpjitterbuffer. After
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* the packets are released from the jitterbuffer, they will be forwarded to an
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* rtpptdemuxer element. The rtpptdemuxer element will demux the packets based
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* Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
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* manager and after being validated forwarded on gstrtpssrcdemuxer element. Each
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* RTP stream is demuxed based on the SSRC and send to a gstrtpjitterbuffer. After
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* the packets are released from the jitterbuffer, they will be forwarded to a
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* gstrtpptdemuxer element. The gstrtpptdemuxer element will demux the packets based
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* on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
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* rtpbin with the session number, SSRC and payload type respectively as the pad
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* gstrtpbin with the session number, SSRC and payload type respectively as the pad
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* name.
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* </para>
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* <para>
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* To also use rtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
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* To also use gstrtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
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* session number must be specified in the pad name.
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* </para>
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* <para>
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@ -55,7 +55,7 @@
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* in the session.
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* </para>
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* <para>
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* To use rtpbin as a sender, request a send_rtp_sink_%%d pad, which will
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* To use gstrtpbin as a sender, request a send_rtp_sink_%%d pad, which will
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* automatically create a send_rtp_src_%%d pad. The session number must be specified when
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* requesting the sink pad. The session manager will modify the
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* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
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@ -71,13 +71,13 @@
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* <para>
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* <programlisting>
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* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
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* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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* gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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* </programlisting>
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* Receive RTP data from port 5000 and send to the session 0 in rtpbin.
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* Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-05-23 (0.10.6)
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* Last reviewed on 2007-05-28 (0.10.5)
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*/
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#ifdef HAVE_CONFIG_H
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@ -265,10 +265,10 @@ create_session (GstRTPBin * rtpbin, gint id)
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GstRTPBinSession *sess;
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GstElement *session, *demux;
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if (!(session = gst_element_factory_make ("rtpsession", NULL)))
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if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
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goto no_session;
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if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
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if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
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goto no_demux;
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sess = g_new0 (GstRTPBinSession, 1);
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@ -294,13 +294,13 @@ create_session (GstRTPBin * rtpbin, gint id)
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/* ERRORS */
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no_session:
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{
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g_warning ("rtpbin: could not create rtpsession element");
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g_warning ("gstrtpbin: could not create gstrtpsession element");
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return NULL;
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}
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no_demux:
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{
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gst_object_unref (session);
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g_warning ("rtpbin: could not create rtpssrcdemux element");
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g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
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return NULL;
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}
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}
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@ -414,10 +414,10 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
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GstElement *buffer, *demux;
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GstRTPBinStream *stream;
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if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
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if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
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goto no_jitterbuffer;
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if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
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if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
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goto no_demux;
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stream = g_new0 (GstRTPBinStream, 1);
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@ -448,13 +448,13 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
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/* ERRORS */
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no_jitterbuffer:
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{
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g_warning ("rtpbin: could not create rtpjitterbuffer element");
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g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
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return NULL;
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}
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no_demux:
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{
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gst_object_unref (buffer);
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g_warning ("rtpbin: could not create rtpptdemux element");
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g_warning ("gstrtpbin: could not create gstrtpptdemux element");
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return NULL;
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}
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}
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@ -834,7 +834,7 @@ create_recv_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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/* ERRORS */
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no_name:
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{
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g_warning ("rtpbin: invalid name given");
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g_warning ("gstrtpbin: invalid name given");
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return NULL;
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}
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create_error:
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@ -844,17 +844,18 @@ create_error:
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}
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existed:
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{
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g_warning ("rtpbin: recv_rtp pad already requested for session %d", sessid);
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g_warning ("gstrtpbin: recv_rtp pad already requested for session %d",
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sessid);
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return NULL;
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}
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pad_failed:
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{
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g_warning ("rtpbin: failed to get session pad");
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g_warning ("gstrtpbin: failed to get session pad");
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return NULL;
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}
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link_failed:
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{
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g_warning ("rtpbin: failed to link pads");
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g_warning ("gstrtpbin: failed to link pads");
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return NULL;
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}
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}
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@ -929,7 +930,7 @@ create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
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/* ERRORS */
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no_name:
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{
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g_warning ("rtpbin: invalid name given");
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g_warning ("gstrtpbin: invalid name given");
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return NULL;
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}
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create_error:
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@ -939,19 +940,19 @@ create_error:
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}
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existed:
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{
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g_warning ("rtpbin: recv_rtcp pad already requested for session %d",
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g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
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sessid);
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return NULL;
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}
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pad_failed:
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{
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g_warning ("rtpbin: failed to get session pad");
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g_warning ("gstrtpbin: failed to get session pad");
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return NULL;
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}
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#if 0
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link_failed:
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{
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g_warning ("rtpbin: failed to link pads");
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g_warning ("gstrtpbin: failed to link pads");
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return NULL;
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}
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#endif
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@ -1018,7 +1019,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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/* ERRORS */
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no_name:
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{
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g_warning ("rtpbin: invalid name given");
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g_warning ("gstrtpbin: invalid name given");
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return NULL;
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}
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create_error:
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@ -1028,17 +1029,19 @@ create_error:
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}
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existed:
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{
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g_warning ("rtpbin: send_rtp pad already requested for session %d", sessid);
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g_warning ("gstrtpbin: send_rtp pad already requested for session %d",
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sessid);
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return NULL;
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}
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pad_failed:
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{
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g_warning ("rtpbin: failed to get session pad for session %d", sessid);
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g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
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return NULL;
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}
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no_srcpad:
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{
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g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
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g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
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sessid);
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return NULL;
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}
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}
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@ -1082,23 +1085,23 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
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/* ERRORS */
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no_name:
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{
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g_warning ("rtpbin: invalid name given");
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g_warning ("gstrtpbin: invalid name given");
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return NULL;
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}
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no_session:
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{
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g_warning ("rtpbin: session with id %d does not exist", sessid);
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g_warning ("gstrtpbin: session with id %d does not exist", sessid);
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return NULL;
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}
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existed:
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{
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g_warning ("rtpbin: send_rtcp_src pad already requested for session %d",
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g_warning ("gstrtpbin: send_rtcp_src pad already requested for session %d",
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sessid);
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return NULL;
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}
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pad_failed:
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{
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g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
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g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
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return NULL;
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}
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}
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@ -1144,7 +1147,7 @@ gst_rtp_bin_request_new_pad (GstElement * element,
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wrong_template:
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{
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GST_RTP_BIN_UNLOCK (rtpbin);
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g_warning ("rtpbin: this is not our template");
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g_warning ("gstrtpbin: this is not our template");
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return NULL;
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}
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}
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@ -18,9 +18,9 @@
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*/
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/**
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* SECTION:element-rtpclient
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* SECTION:element-gstrtpclient
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* @short_description: handle media from one RTP client
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* @see_also: rtpjitterbuffer, rtpbin, rtpsession
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* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpsession
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*
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* <refsect2>
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* <para>
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@ -38,7 +38,7 @@
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-04-02 (0.10.6)
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* Last reviewed on 2007-04-02 (0.10.5)
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*/
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#ifdef HAVE_CONFIG_H
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@ -136,11 +136,11 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
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stream->ssrc = ssrc;
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stream->client = rtpclient;
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stream->jitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL);
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stream->jitterbuffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL);
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if (!stream->jitterbuffer)
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goto no_jitterbuffer;
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stream->ptdemux = gst_element_factory_make ("rtpptdemux", NULL);
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stream->ptdemux = gst_element_factory_make ("gstrtpptdemux", NULL);
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if (!stream->ptdemux)
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goto no_ptdemux;
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@ -180,14 +180,14 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
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no_jitterbuffer:
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{
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g_free (stream);
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g_warning ("could not create rtpjitterbuffer element");
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g_warning ("gstrtpclient: could not create gstrtpjitterbuffer element");
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return NULL;
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}
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no_ptdemux:
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{
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gst_object_unref (stream->jitterbuffer);
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g_free (stream);
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g_warning ("could not create rtpptdemux element");
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g_warning ("gstrtpclient: could not create gstrtpptdemux element");
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return NULL;
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}
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could_not_link:
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@ -195,7 +195,7 @@ could_not_link:
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gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
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gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux);
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g_free (stream);
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g_warning ("could not link jitterbuffer and rtpptdemux element");
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g_warning ("gstrtpclient: could not link jitterbuffer and ptdemux element");
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return NULL;
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}
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}
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@ -455,27 +455,27 @@ gst_rtp_client_request_new_pad (GstElement * element,
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/* ERRORS */
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wrong_direction:
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{
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g_warning ("rtpclient: request pad that is not a SINK pad");
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g_warning ("gstrtpclient: request pad that is not a SINK pad");
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return NULL;
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}
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wrong_template:
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{
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g_warning ("rtpclient: this is not our template");
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g_warning ("gstrtpclient: this is not our template");
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return NULL;
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}
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no_name:
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{
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g_warning ("rtpclient: no padname was specified");
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g_warning ("gstrtpclient: no padname was specified");
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return NULL;
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}
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stream_exists:
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{
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g_warning ("rtpclient: stream with SSRC %d already registered", ssrc);
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g_warning ("gstrtpclient: stream with SSRC %d already registered", ssrc);
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return NULL;
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}
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stream_not_found:
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{
|
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g_warning ("rtpclient: stream with SSRC %d not yet registered", ssrc);
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g_warning ("gstrtpclient: stream with SSRC %d not yet registered", ssrc);
|
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return NULL;
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}
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}
|
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|
|
|
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|
|
@ -24,7 +24,7 @@
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*/
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/**
|
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* SECTION:element-rtpjitterbuffer
|
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* SECTION:element-gstrtpjitterbuffer
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* @short_description: buffer, reorder and remove duplicate RTP packets to
|
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* compensate for network oddities.
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*
|
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|
|
@ -33,7 +33,7 @@
|
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* This element reorders and removes duplicate RTP packets as they are received
|
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* from a network source. It will also wait for missing packets up to a
|
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* configurable time limit using the ::latency property. Packets arriving too
|
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* late are considered as lost packets.
|
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* late are considered to be lost packets.
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* </para>
|
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* <para>
|
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* This element acts as a live element and so adds ::latency to the pipeline.
|
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|
|
@ -45,12 +45,12 @@
|
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* previous pt-map use the ::clear-pt-map signal.
|
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* </para>
|
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* <para>
|
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* This element will automatically be used inside rtpbin.
|
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* This element will automatically be used inside gstrtpbin.
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* </para>
|
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* <title>Example pipelines</title>
|
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* <para>
|
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* <programlisting>
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* gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
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* gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
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* </programlisting>
|
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* Connect to a streaming server and decode the MPEG video. The jitterbuffer is
|
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* inserted into the pipeline to smooth out network jitter and to reorder the
|
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|
|
@ -58,7 +58,7 @@
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* </para>
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* </refsect2>
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*
|
||||
* Last reviewed on 2007-05-22 (0.10.6)
|
||||
* Last reviewed on 2007-05-28 (0.10.5)
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
|
|
|
|||
|
|
@ -31,26 +31,27 @@
|
|||
static gboolean
|
||||
plugin_init (GstPlugin * plugin)
|
||||
{
|
||||
if (!gst_element_register (plugin, "rtpbin", GST_RANK_NONE, GST_TYPE_RTP_BIN))
|
||||
if (!gst_element_register (plugin, "gstrtpbin", GST_RANK_NONE,
|
||||
GST_TYPE_RTP_BIN))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_element_register (plugin, "rtpclient", GST_RANK_NONE,
|
||||
if (!gst_element_register (plugin, "gstrtpclient", GST_RANK_NONE,
|
||||
GST_TYPE_RTP_CLIENT))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_element_register (plugin, "rtpjitterbuffer", GST_RANK_NONE,
|
||||
if (!gst_element_register (plugin, "gstrtpjitterbuffer", GST_RANK_NONE,
|
||||
GST_TYPE_RTP_JITTER_BUFFER))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_element_register (plugin, "rtpptdemux", GST_RANK_NONE,
|
||||
if (!gst_element_register (plugin, "gstrtpptdemux", GST_RANK_NONE,
|
||||
GST_TYPE_RTP_PT_DEMUX))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_element_register (plugin, "rtpsession", GST_RANK_NONE,
|
||||
if (!gst_element_register (plugin, "gstrtpsession", GST_RANK_NONE,
|
||||
GST_TYPE_RTP_SESSION))
|
||||
return FALSE;
|
||||
|
||||
if (!gst_element_register (plugin, "rtpssrcdemux", GST_RANK_NONE,
|
||||
if (!gst_element_register (plugin, "gstrtpssrcdemux", GST_RANK_NONE,
|
||||
GST_TYPE_RTP_SSRC_DEMUX))
|
||||
return FALSE;
|
||||
|
||||
|
|
@ -59,6 +60,6 @@ plugin_init (GstPlugin * plugin)
|
|||
|
||||
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
|
||||
GST_VERSION_MINOR,
|
||||
"rtpmanager",
|
||||
"gstrtpmanager",
|
||||
"RTP session management plugin library",
|
||||
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
|
||||
|
|
|
|||
|
|
@ -24,12 +24,12 @@
|
|||
*/
|
||||
|
||||
/**
|
||||
* SECTION:element-rtpptdemux
|
||||
* SECTION:element-gstrtpptdemux
|
||||
* @short_description: separate RTP payloads based on the payload type
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
* rtpptdemux acts as a demuxer for RTP packets based on the payload type of the
|
||||
* gstrtpptdemux 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.
|
||||
* </para>
|
||||
|
|
@ -45,14 +45,14 @@
|
|||
* <title>Example pipelines</title>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink
|
||||
* gst-launch udpsrc caps="application/x-rtp" ! gstrtpptdemux ! fakesink
|
||||
* </programlisting>
|
||||
* Takes an RTP stream and send the RTP packets with the first detected payload
|
||||
* type to fakesink, discarding the other payload types.
|
||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* Last reviewed on 2007-05-22 (0.10.6)
|
||||
* Last reviewed on 2007-05-28 (0.10.5)
|
||||
*/
|
||||
|
||||
/*
|
||||
|
|
|
|||
|
|
@ -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
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
|
|
@ -47,20 +47,20 @@
|
|||
* </itemizedlist>
|
||||
* </para>
|
||||
* <para>
|
||||
* 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.
|
||||
* </para>
|
||||
* <para>
|
||||
* 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.
|
||||
* </para>
|
||||
* <para>
|
||||
* 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.
|
||||
* </para>
|
||||
* <para>
|
||||
* 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 @@
|
|||
* <title>Example pipelines</title>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* 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
|
||||
* </programlisting>
|
||||
* 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 @@
|
|||
* </para>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* 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
|
||||
* </programlisting>
|
||||
|
|
@ -108,24 +108,24 @@
|
|||
* </para>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* 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
|
||||
* </programlisting>
|
||||
* Send theora RTP packets through the session manager and out on UDP port 5000.
|
||||
* </para>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* 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
|
||||
* </programlisting>
|
||||
* 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.
|
||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* 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;
|
||||
}
|
||||
}
|
||||
|
|
|
|||
|
|
@ -20,12 +20,12 @@
|
|||
*/
|
||||
|
||||
/**
|
||||
* SECTION:element-rtpssrcdemux
|
||||
* SECTION:element-gstrtpssrcdemux
|
||||
* @short_description: separate RTP payloads based on the SSRC
|
||||
*
|
||||
* <refsect2>
|
||||
* <para>
|
||||
* rtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
|
||||
* gstrtpssrcdemux 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.
|
||||
* </para>
|
||||
|
|
@ -36,14 +36,14 @@
|
|||
* <title>Example pipelines</title>
|
||||
* <para>
|
||||
* <programlisting>
|
||||
* gst-launch udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink
|
||||
* gst-launch udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
|
||||
* </programlisting>
|
||||
* Takes an RTP stream and send the RTP packets with the first detected SSRC
|
||||
* to fakesink, discarding the other SSRCs.
|
||||
* </para>
|
||||
* </refsect2>
|
||||
*
|
||||
* Last reviewed on 2007-05-23 (0.10.6)
|
||||
* Last reviewed on 2007-05-28 (0.10.5)
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
|
|
|
|||
Loading…
Reference in a new issue