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Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.

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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 2007-05-28 16:37:47 +00:00 committed by Tim-Philipp Müller
parent 2a8cfc6410
commit 0c4fe985b6
7 changed files with 97 additions and 93 deletions
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89
gst/rtpmanager/gstrtpbin.c
View file

@ -18,34 +18,34 @@
*/ */
/** /**
* SECTION:element-rtpbin * SECTION:element-gstrtpbin
* @short_description: handle media from one RTP bin * @short_description: handle media from one RTP bin
* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
* *
* <refsect2> * <refsect2>
* <para> * <para>
* RTP bin combines the functions of rtpsession, rtpssrcdemux, rtpjitterbuffer * RTP bin combines the functions of gstrtpsession, gstrtpssrcdemux, gstrtpjitterbuffer
* and rtpptdemux in one element. It allows for multiple rtpsessions that will * and gstrtpptdemux in one element. It allows for multiple RTP sessions that will
* be synchronized together using RTCP SR packets. * be synchronized together using RTCP SR packets.
* </para> * </para>
* <para> * <para>
* rtpbin is configured with a number of request pads that define the * gstrtpbin is configured with a number of request pads that define the
* functionality that is activated, similar to the rtpsession element. * functionality that is activated, similar to the gstrtpsession element.
* </para> * </para>
* <para> * <para>
* To use rtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session * To use gstrtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
* number must be specified in the pad name. * number must be specified in the pad name.
* Data received on the recv_rtp_sink_%%d pad will be processed in the rtpsession * Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
* manager and after being validated forwarded on rtpssrcdemuxer element. Each * manager and after being validated forwarded on gstrtpssrcdemuxer element. Each
* RTP stream is demuxed based on the SSRC and send to a rtpjitterbuffer. After * RTP stream is demuxed based on the SSRC and send to a gstrtpjitterbuffer. After
* the packets are released from the jitterbuffer, they will be forwarded to an * the packets are released from the jitterbuffer, they will be forwarded to a
* rtpptdemuxer element. The rtpptdemuxer element will demux the packets based * gstrtpptdemuxer element. The gstrtpptdemuxer element will demux the packets based
* on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on * on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
* rtpbin with the session number, SSRC and payload type respectively as the pad * gstrtpbin with the session number, SSRC and payload type respectively as the pad
* name. * name.
* </para> * </para>
* <para> * <para>
* To also use rtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The * To also use gstrtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
* session number must be specified in the pad name. * session number must be specified in the pad name.
* </para> * </para>
* <para> * <para>
@ -55,7 +55,7 @@
* in the session. * in the session.
* </para> * </para>
* <para> * <para>
* To use rtpbin as a sender, request a send_rtp_sink_%%d pad, which will * To use gstrtpbin as a sender, request a send_rtp_sink_%%d pad, which will
* automatically create a send_rtp_src_%%d pad. The session number must be specified when * automatically create a send_rtp_src_%%d pad. The session number must be specified when
* requesting the sink pad. The session manager will modify the * requesting the sink pad. The session manager will modify the
* SSRC in the RTP packets to its own SSRC and wil forward the packets on the * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
@ -71,13 +71,13 @@
* <para> * <para>
* <programlisting> * <programlisting>
* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \ * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink * gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* </programlisting> * </programlisting>
* Receive RTP data from port 5000 and send to the session 0 in rtpbin. * Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
* </para> * </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-05-23 (0.10.6) * Last reviewed on 2007-05-28 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
@ -265,10 +265,10 @@ create_session (GstRTPBin * rtpbin, gint id)
GstRTPBinSession *sess; GstRTPBinSession *sess;
GstElement *session, *demux; GstElement *session, *demux;
if (!(session = gst_element_factory_make ("rtpsession", NULL))) if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
goto no_session; goto no_session;
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL))) if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
goto no_demux; goto no_demux;
sess = g_new0 (GstRTPBinSession, 1); sess = g_new0 (GstRTPBinSession, 1);
@ -294,13 +294,13 @@ create_session (GstRTPBin * rtpbin, gint id)
/* ERRORS */ /* ERRORS */
no_session: no_session:
{ {
g_warning ("rtpbin: could not create rtpsession element"); g_warning ("gstrtpbin: could not create gstrtpsession element");
return NULL; return NULL;
} }
no_demux: no_demux:
{ {
gst_object_unref (session); gst_object_unref (session);
g_warning ("rtpbin: could not create rtpssrcdemux element"); g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
return NULL; return NULL;
} }
} }
@ -414,10 +414,10 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
GstElement *buffer, *demux; GstElement *buffer, *demux;
GstRTPBinStream *stream; GstRTPBinStream *stream;
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL))) if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
goto no_jitterbuffer; goto no_jitterbuffer;
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL))) if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
goto no_demux; goto no_demux;
stream = g_new0 (GstRTPBinStream, 1); stream = g_new0 (GstRTPBinStream, 1);
@ -448,13 +448,13 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
/* ERRORS */ /* ERRORS */
no_jitterbuffer: no_jitterbuffer:
{ {
g_warning ("rtpbin: could not create rtpjitterbuffer element"); g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
return NULL; return NULL;
} }
no_demux: no_demux:
{ {
gst_object_unref (buffer); gst_object_unref (buffer);
g_warning ("rtpbin: could not create rtpptdemux element"); g_warning ("gstrtpbin: could not create gstrtpptdemux element");
return NULL; return NULL;
} }
} }
@ -834,7 +834,7 @@ create_recv_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */ /* ERRORS */
no_name: no_name:
{ {
g_warning ("rtpbin: invalid name given"); g_warning ("gstrtpbin: invalid name given");
return NULL; return NULL;
} }
create_error: create_error:
@ -844,17 +844,18 @@ create_error:
} }
existed: existed:
{ {
g_warning ("rtpbin: recv_rtp pad already requested for session %d", sessid); g_warning ("gstrtpbin: recv_rtp pad already requested for session %d",
sessid);
return NULL; return NULL;
} }
pad_failed: pad_failed:
{ {
g_warning ("rtpbin: failed to get session pad"); g_warning ("gstrtpbin: failed to get session pad");
return NULL; return NULL;
} }
link_failed: link_failed:
{ {
g_warning ("rtpbin: failed to link pads"); g_warning ("gstrtpbin: failed to link pads");
return NULL; return NULL;
} }
} }
@ -929,7 +930,7 @@ create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
/* ERRORS */ /* ERRORS */
no_name: no_name:
{ {
g_warning ("rtpbin: invalid name given"); g_warning ("gstrtpbin: invalid name given");
return NULL; return NULL;
} }
create_error: create_error:
@ -939,19 +940,19 @@ create_error:
} }
existed: existed:
{ {
g_warning ("rtpbin: recv_rtcp pad already requested for session %d", g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
sessid); sessid);
return NULL; return NULL;
} }
pad_failed: pad_failed:
{ {
g_warning ("rtpbin: failed to get session pad"); g_warning ("gstrtpbin: failed to get session pad");
return NULL; return NULL;
} }
#if 0 #if 0
link_failed: link_failed:
{ {
g_warning ("rtpbin: failed to link pads"); g_warning ("gstrtpbin: failed to link pads");
return NULL; return NULL;
} }
#endif #endif
@ -1018,7 +1019,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */ /* ERRORS */
no_name: no_name:
{ {
g_warning ("rtpbin: invalid name given"); g_warning ("gstrtpbin: invalid name given");
return NULL; return NULL;
} }
create_error: create_error:
@ -1028,17 +1029,19 @@ create_error:
} }
existed: existed:
{ {
g_warning ("rtpbin: send_rtp pad already requested for session %d", sessid); g_warning ("gstrtpbin: send_rtp pad already requested for session %d",
sessid);
return NULL; return NULL;
} }
pad_failed: pad_failed:
{ {
g_warning ("rtpbin: failed to get session pad for session %d", sessid); g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
return NULL; return NULL;
} }
no_srcpad: no_srcpad:
{ {
g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid); g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
sessid);
return NULL; return NULL;
} }
} }
@ -1082,23 +1085,23 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */ /* ERRORS */
no_name: no_name:
{ {
g_warning ("rtpbin: invalid name given"); g_warning ("gstrtpbin: invalid name given");
return NULL; return NULL;
} }
no_session: no_session:
{ {
g_warning ("rtpbin: session with id %d does not exist", sessid); g_warning ("gstrtpbin: session with id %d does not exist", sessid);
return NULL; return NULL;
} }
existed: existed:
{ {
g_warning ("rtpbin: send_rtcp_src pad already requested for session %d", g_warning ("gstrtpbin: send_rtcp_src pad already requested for session %d",
sessid); sessid);
return NULL; return NULL;
} }
pad_failed: pad_failed:
{ {
g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid); g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
return NULL; return NULL;
} }
} }
@ -1144,7 +1147,7 @@ gst_rtp_bin_request_new_pad (GstElement * element,
wrong_template: wrong_template:
{ {
GST_RTP_BIN_UNLOCK (rtpbin); GST_RTP_BIN_UNLOCK (rtpbin);
g_warning ("rtpbin: this is not our template"); g_warning ("gstrtpbin: this is not our template");
return NULL; return NULL;
} }
} }

26
gst/rtpmanager/gstrtpclient.c
View file

@ -18,9 +18,9 @@
*/ */
/** /**
* SECTION:element-rtpclient * SECTION:element-gstrtpclient
* @short_description: handle media from one RTP client * @short_description: handle media from one RTP client
* @see_also: rtpjitterbuffer, rtpbin, rtpsession * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpsession
* *
* <refsect2> * <refsect2>
* <para> * <para>
@ -38,7 +38,7 @@
* </para> * </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-04-02 (0.10.6) * Last reviewed on 2007-04-02 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
@ -136,11 +136,11 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
stream->ssrc = ssrc; stream->ssrc = ssrc;
stream->client = rtpclient; stream->client = rtpclient;
stream->jitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL); stream->jitterbuffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL);
if (!stream->jitterbuffer) if (!stream->jitterbuffer)
goto no_jitterbuffer; goto no_jitterbuffer;
stream->ptdemux = gst_element_factory_make ("rtpptdemux", NULL); stream->ptdemux = gst_element_factory_make ("gstrtpptdemux", NULL);
if (!stream->ptdemux) if (!stream->ptdemux)
goto no_ptdemux; goto no_ptdemux;
@ -180,14 +180,14 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
no_jitterbuffer: no_jitterbuffer:
{ {
g_free (stream); g_free (stream);
g_warning ("could not create rtpjitterbuffer element"); g_warning ("gstrtpclient: could not create gstrtpjitterbuffer element");
return NULL; return NULL;
} }
no_ptdemux: no_ptdemux:
{ {
gst_object_unref (stream->jitterbuffer); gst_object_unref (stream->jitterbuffer);
g_free (stream); g_free (stream);
g_warning ("could not create rtpptdemux element"); g_warning ("gstrtpclient: could not create gstrtpptdemux element");
return NULL; return NULL;
} }
could_not_link: could_not_link:
@ -195,7 +195,7 @@ could_not_link:
gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer); gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux); gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux);
g_free (stream); g_free (stream);
g_warning ("could not link jitterbuffer and rtpptdemux element"); g_warning ("gstrtpclient: could not link jitterbuffer and ptdemux element");
return NULL; return NULL;
} }
} }
@ -455,27 +455,27 @@ gst_rtp_client_request_new_pad (GstElement * element,
/* ERRORS */ /* ERRORS */
wrong_direction: wrong_direction:
{ {
g_warning ("rtpclient: request pad that is not a SINK pad"); g_warning ("gstrtpclient: request pad that is not a SINK pad");
return NULL; return NULL;
} }
wrong_template: wrong_template:
{ {
g_warning ("rtpclient: this is not our template"); g_warning ("gstrtpclient: this is not our template");
return NULL; return NULL;
} }
no_name: no_name:
{ {
g_warning ("rtpclient: no padname was specified"); g_warning ("gstrtpclient: no padname was specified");
return NULL; return NULL;
} }
stream_exists: stream_exists:
{ {
g_warning ("rtpclient: stream with SSRC %d already registered", ssrc); g_warning ("gstrtpclient: stream with SSRC %d already registered", ssrc);
return NULL; return NULL;
} }
stream_not_found: stream_not_found:
{ {
g_warning ("rtpclient: stream with SSRC %d not yet registered", ssrc); g_warning ("gstrtpclient: stream with SSRC %d not yet registered", ssrc);
return NULL; return NULL;
} }
} }

10
gst/rtpmanager/gstrtpjitterbuffer.c
View file

@ -24,7 +24,7 @@
*/ */
/** /**
* SECTION:element-rtpjitterbuffer * SECTION:element-gstrtpjitterbuffer
* @short_description: buffer, reorder and remove duplicate RTP packets to * @short_description: buffer, reorder and remove duplicate RTP packets to
* compensate for network oddities. * compensate for network oddities.
* *
@ -33,7 +33,7 @@
* 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
* configurable time limit using the ::latency property. Packets arriving too * configurable time limit using the ::latency property. Packets arriving too
* late are considered as lost packets. * late are considered to be lost packets.
* </para> * </para>
* <para> * <para>
* This element acts as a live element and so adds ::latency to the pipeline. * This element acts as a live element and so adds ::latency to the pipeline.
@ -45,12 +45,12 @@
* previous pt-map use the ::clear-pt-map signal. * previous pt-map use the ::clear-pt-map signal.
* </para> * </para>
* <para> * <para>
* This element will automatically be used inside rtpbin. * This element will automatically be used inside gstrtpbin.
* </para> * </para>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * <para>
* <programlisting> * <programlisting>
* gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
* </programlisting> * </programlisting>
* 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
@ -58,7 +58,7 @@
* </para> * </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-05-22 (0.10.6) * Last reviewed on 2007-05-28 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

15
gst/rtpmanager/gstrtpmanager.c
View file

@ -31,26 +31,27 @@
static gboolean static gboolean
plugin_init (GstPlugin * plugin) 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; return FALSE;
if (!gst_element_register (plugin, "rtpclient", GST_RANK_NONE, if (!gst_element_register (plugin, "gstrtpclient", GST_RANK_NONE,
GST_TYPE_RTP_CLIENT)) GST_TYPE_RTP_CLIENT))
return FALSE; 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)) GST_TYPE_RTP_JITTER_BUFFER))
return FALSE; 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)) GST_TYPE_RTP_PT_DEMUX))
return FALSE; return FALSE;
if (!gst_element_register (plugin, "rtpsession", GST_RANK_NONE, if (!gst_element_register (plugin, "gstrtpsession", GST_RANK_NONE,
GST_TYPE_RTP_SESSION)) GST_TYPE_RTP_SESSION))
return FALSE; 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)) GST_TYPE_RTP_SSRC_DEMUX))
return FALSE; return FALSE;
@ -59,6 +60,6 @@ plugin_init (GstPlugin * plugin)
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR, GST_VERSION_MINOR,
"rtpmanager", "gstrtpmanager",
"RTP session management plugin library", "RTP session management plugin library",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN) plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

8
gst/rtpmanager/gstrtpptdemux.c
View file

@ -24,12 +24,12 @@
*/ */
/** /**
* SECTION:element-rtpptdemux * SECTION:element-gstrtpptdemux
* @short_description: separate RTP payloads based on the payload type * @short_description: separate RTP payloads based on the payload type
* *
* <refsect2> * <refsect2>
* <para> * <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 * packets. Its main purpose is to allow an application to easily receive and
* decode an RTP stream with multiple payload types. * decode an RTP stream with multiple payload types.
* </para> * </para>
@ -45,14 +45,14 @@
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * <para>
* <programlisting> * <programlisting>
* gst-launch udpsrc caps="application/x-rtp" ! rtpptdemux ! fakesink * gst-launch udpsrc caps="application/x-rtp" ! gstrtpptdemux ! fakesink
* </programlisting> * </programlisting>
* Takes an RTP stream and send the RTP packets with the first detected payload * Takes an RTP stream and send the RTP packets with the first detected payload
* type to fakesink, discarding the other payload types. * type to fakesink, discarding the other payload types.
* </para> * </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-05-22 (0.10.6) * Last reviewed on 2007-05-28 (0.10.5)
*/ */
/* /*

34
gst/rtpmanager/gstrtpsession.c
View file

@ -18,9 +18,9 @@
*/ */
/** /**
* SECTION:element-rtpsession * SECTION:element-gstrtpsession
* @short_description: an RTP session manager * @short_description: an RTP session manager
* @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
* *
* <refsect2> * <refsect2>
* <para> * <para>
@ -47,20 +47,20 @@
* </itemizedlist> * </itemizedlist>
* </para> * </para>
* <para> * <para>
* The rtpsession will not demux packets based on SSRC or payload type, nor will * The gstrtpsession will not demux packets based on SSRC or payload type, nor will
* it correct for packet reordering and jitter. Use rtpssrcdemux, rtpptdemux and * it correct for packet reordering and jitter. Use gstrtpssrcdemux, gstrtpptdemux and
* rtpjitterbuffer in addition to rtpsession to perform these tasks. It is * gstrtpjitterbuffer in addition to gstrtpsession to perform these tasks. It is
* usually a good idea to use rtpbin, which combines all these features in one * usually a good idea to use gstrtpbin, which combines all these features in one
* element. * element.
* </para> * </para>
* <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 * 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 * will be processed in the session and after being validated forwarded on the
* recv_rtp_src pad. * recv_rtp_src pad.
* </para> * </para>
* <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 * 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 * 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 * 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. * that should be sent to all participants in the session.
* </para> * </para>
* <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 * 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 * 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. * send_rtp_src pad after updating its internal state.
@ -86,7 +86,7 @@
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * <para>
* <programlisting> * <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> * </programlisting>
* 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
@ -95,7 +95,7 @@
* </para> * </para>
* <para> * <para>
* <programlisting> * <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 \ * .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
* </programlisting> * </programlisting>
@ -108,24 +108,24 @@
* </para> * </para>
* <para> * <para>
* <programlisting> * <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> * </programlisting>
* Send theora RTP packets through the session manager and out on UDP port 5000. * Send theora RTP packets through the session manager and out on UDP port 5000.
* </para> * </para>
* <para> * <para>
* <programlisting> * <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 * ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
* </programlisting> * </programlisting>
* Send theora RTP packets through the session manager and out on UDP port 5000. * 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 * correctly because the second udpsink will not preroll correctly (no RTCP
* packets are sent in the PAUSED state). Applications should manually set and * 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. * keep (see #gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
* </para> * </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-05-23 (0.10.6) * Last reviewed on 2007-05-28 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
@ -1001,13 +1001,13 @@ gst_rtp_session_request_new_pad (GstElement * element,
wrong_template: wrong_template:
{ {
GST_RTP_SESSION_UNLOCK (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("rtpsession: this is not our template"); g_warning ("gstrtpsession: this is not our template");
return NULL; return NULL;
} }
exists: exists:
{ {
GST_RTP_SESSION_UNLOCK (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("rtpsession: pad already requested"); g_warning ("gstrtpsession: pad already requested");
return NULL; return NULL;
} }
} }

8
gst/rtpmanager/gstrtpssrcdemux.c
View file

@ -20,12 +20,12 @@
*/ */
/** /**
* SECTION:element-rtpssrcdemux * SECTION:element-gstrtpssrcdemux
* @short_description: separate RTP payloads based on the SSRC * @short_description: separate RTP payloads based on the SSRC
* *
* <refsect2> * <refsect2>
* <para> * <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 * 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.
* </para> * </para>
@ -36,14 +36,14 @@
* <title>Example pipelines</title> * <title>Example pipelines</title>
* <para> * <para>
* <programlisting> * <programlisting>
* gst-launch udpsrc caps="application/x-rtp" ! rtpssrcdemux ! fakesink * gst-launch udpsrc caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
* </programlisting> * </programlisting>
* 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.
* </para> * </para>
* </refsect2> * </refsect2>
* *
* Last reviewed on 2007-05-23 (0.10.6) * Last reviewed on 2007-05-28 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H