mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-27 20:21:24 +00:00
Code Issues 1 Projects Releases Wiki Activity

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

Browse source 
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
parent 6587432049
commit 3a496fd7eb
10 changed files with 134 additions and 111 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

19
ChangeLog
View file

@ -1,3 +1,22 @@
2007-05-28 Wim Taymans <wim@fluendo.com>
* 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.
2007-05-24 Tim-Philipp Müller <tim at centricular dot net>
Patch by: Vincent Torri <vtorri at univ-evry fr>

12
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -22,11 +22,11 @@
<xi:include href="xml/element-rganalysis.xml" />
<xi:include href="xml/element-rglimiter.xml" />
<xi:include href="xml/element-rgvolume.xml" />
<xi:include href="xml/element-rtpbin.xml" />
<xi:include href="xml/element-rtpjitterbuffer.xml" />
<xi:include href="xml/element-rtpptdemux.xml" />
<xi:include href="xml/element-rtpsession.xml" />
<xi:include href="xml/element-rtpssrcdemux.xml" />
<xi:include href="xml/element-gstrtpbin.xml" />
<xi:include href="xml/element-gstrtpjitterbuffer.xml" />
<xi:include href="xml/element-gstrtpptdemux.xml" />
<xi:include href="xml/element-gstrtpsession.xml" />
<xi:include href="xml/element-gstrtpssrcdemux.xml" />
<xi:include href="xml/element-sdlaudiosink.xml" />
<xi:include href="xml/element-sdlvideosink.xml" />
<xi:include href="xml/element-trm.xml" />
@ -58,7 +58,7 @@
<xi:include href="xml/plugin-osxvideo.xml" />
<xi:include href="xml/plugin-qtdemux.xml" />
<xi:include href="xml/plugin-replaygain.xml" />
<xi:include href="xml/plugin-rtpmanager.xml" />
<xi:include href="xml/plugin-gstrtpmanager.xml" />
<xi:include href="xml/plugin-sdl.xml" />
<xi:include href="xml/plugin-spectrum.xml" />
<xi:include href="xml/plugin-speed.xml" />

24
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -80,9 +80,9 @@ GstRgVolumeClass
</SECTION>
<SECTION>
<FILE>element-rtpbin</FILE>
<FILE>element-gstrtpbin</FILE>
GstRTPBin
<TITLE>rtpbin</TITLE>
<TITLE>gstrtpbin</TITLE>
<SUBSECTION Standard>
GstRTPBinPrivate
GstRTPBinClass
@ -95,8 +95,8 @@ GST_IS_RTP_BIN_CLASS
</SECTION>
<SECTION>
<FILE>element-rtpclient</FILE>
<TITLE>rtpclient</TITLE>
<FILE>element-gstrtpclient</FILE>
<TITLE>gstrtpclient</TITLE>
GstRTPClient
<SUBSECTION Standard>
GstRTPClientClass
@ -110,8 +110,8 @@ GST_IS_RTP_CLIENT_CLASS
</SECTION>
<SECTION>
<FILE>element-rtpjitterbuffer</FILE>
<TITLE>rtpjitterbuffer</TITLE>
<FILE>element-gstrtpjitterbuffer</FILE>
<TITLE>gstrtpjitterbuffer</TITLE>
GstRTPJitterBuffer
<SUBSECTION Standard>
GstRTPJitterBufferClass
@ -125,8 +125,8 @@ GST_IS_RTP_JITTER_BUFFER_CLASS
</SECTION>
<SECTION>
<FILE>element-rtpptdemux</FILE>
<TITLE>rtpptdemux</TITLE>
<FILE>element-gstrtpptdemux</FILE>
<TITLE>gstrtpptdemux</TITLE>
GstRTPPtDemux
<SUBSECTION Standard>
GstRTPPtDemuxClass
@ -140,8 +140,8 @@ GST_IS_RTP_PT_DEMUX_CLASS
</SECTION>
<SECTION>
<FILE>element-rtpsession</FILE>
<TITLE>rtpsession</TITLE>
<FILE>element-gstrtpsession</FILE>
<TITLE>gstrtpsession</TITLE>
GstRTPSession
<SUBSECTION Standard>
GstRTPSessionClass
@ -155,8 +155,8 @@ GST_IS_RTP_SESSION_CLASS
</SECTION>
<SECTION>
<FILE>element-rtpssrcdemux</FILE>
<TITLE>rtpssrcdemux</TITLE>
<FILE>element-gstrtpssrcdemux</FILE>
<TITLE>gstrtpssrcdemux</TITLE>
GstRTPSsrcDemux
<SUBSECTION Standard>
GstRTPSsrcDemuxClass

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
* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
* @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
*
* <refsect2>
* <para>
* RTP bin combines the functions of rtpsession, rtpssrcdemux, rtpjitterbuffer
* and rtpptdemux in one element. It allows for multiple rtpsessions that will
* RTP bin combines the functions of gstrtpsession, gstrtpssrcdemux, gstrtpjitterbuffer
* and gstrtpptdemux in one element. It allows for multiple RTP sessions that will
* be synchronized together using RTCP SR packets.
* </para>
* <para>
* rtpbin is configured with a number of request pads that define the
* functionality that is activated, similar to the rtpsession element.
* gstrtpbin is configured with a number of request pads that define the
* functionality that is activated, similar to the gstrtpsession element.
* </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.
* Data received on the recv_rtp_sink_%%d pad will be processed in the rtpsession
* manager and after being validated forwarded on rtpssrcdemuxer element. Each
* RTP stream is demuxed based on the SSRC and send to a rtpjitterbuffer. After
* the packets are released from the jitterbuffer, they will be forwarded to an
* rtpptdemuxer element. The rtpptdemuxer element will demux the packets based
* Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
* manager and after being validated forwarded on gstrtpssrcdemuxer element. Each
* 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 a
* 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
* 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.
* </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.
* </para>
* <para>
@ -55,7 +55,7 @@
* in the session.
* </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
* 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
@ -71,13 +71,13 @@
* <para>
* <programlisting>
* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
* </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>
* </refsect2>
*
* Last reviewed on 2007-05-23 (0.10.6)
* Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
@ -265,10 +265,10 @@ create_session (GstRTPBin * rtpbin, gint id)
GstRTPBinSession *sess;
GstElement *session, *demux;
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
goto no_session;
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
goto no_demux;
sess = g_new0 (GstRTPBinSession, 1);
@ -294,13 +294,13 @@ create_session (GstRTPBin * rtpbin, gint id)
/* ERRORS */
no_session:
{
g_warning ("rtpbin: could not create rtpsession element");
g_warning ("gstrtpbin: could not create gstrtpsession element");
return NULL;
}
no_demux:
{
gst_object_unref (session);
g_warning ("rtpbin: could not create rtpssrcdemux element");
g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
return NULL;
}
}
@ -414,10 +414,10 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
GstElement *buffer, *demux;
GstRTPBinStream *stream;
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
goto no_jitterbuffer;
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
goto no_demux;
stream = g_new0 (GstRTPBinStream, 1);
@ -448,13 +448,13 @@ create_stream (GstRTPBinSession * session, guint32 ssrc)
/* ERRORS */
no_jitterbuffer:
{
g_warning ("rtpbin: could not create rtpjitterbuffer element");
g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
return NULL;
}
no_demux:
{
gst_object_unref (buffer);
g_warning ("rtpbin: could not create rtpptdemux element");
g_warning ("gstrtpbin: could not create gstrtpptdemux element");
return NULL;
}
}
@ -834,7 +834,7 @@ create_recv_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
@ -844,17 +844,18 @@ create_error:
}
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;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad");
g_warning ("gstrtpbin: failed to get session pad");
return NULL;
}
link_failed:
{
g_warning ("rtpbin: failed to link pads");
g_warning ("gstrtpbin: failed to link pads");
return NULL;
}
}
@ -929,7 +930,7 @@ create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
@ -939,19 +940,19 @@ create_error:
}
existed:
{
g_warning ("rtpbin: recv_rtcp pad already requested for session %d",
g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
sessid);
return NULL;
}
pad_failed:
{
g_warning ("rtpbin: failed to get session pad");
g_warning ("gstrtpbin: failed to get session pad");
return NULL;
}
#if 0
link_failed:
{
g_warning ("rtpbin: failed to link pads");
g_warning ("gstrtpbin: failed to link pads");
return NULL;
}
#endif
@ -1018,7 +1019,7 @@ create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
create_error:
@ -1028,17 +1029,19 @@ create_error:
}
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;
}
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;
}
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;
}
}
@ -1082,23 +1085,23 @@ create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
/* ERRORS */
no_name:
{
g_warning ("rtpbin: invalid name given");
g_warning ("gstrtpbin: invalid name given");
return NULL;
}
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;
}
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);
return NULL;
}
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;
}
}
@ -1144,7 +1147,7 @@ gst_rtp_bin_request_new_pad (GstElement * element,
wrong_template:
{
GST_RTP_BIN_UNLOCK (rtpbin);
g_warning ("rtpbin: this is not our template");
g_warning ("gstrtpbin: this is not our template");
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
* @see_also: rtpjitterbuffer, rtpbin, rtpsession
* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpsession
*
* <refsect2>
* <para>
@ -38,7 +38,7 @@
* </para>
* </refsect2>
*
* Last reviewed on 2007-04-02 (0.10.6)
* Last reviewed on 2007-04-02 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
@ -136,11 +136,11 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
stream->ssrc = ssrc;
stream->client = rtpclient;
stream->jitterbuffer = gst_element_factory_make ("rtpjitterbuffer", NULL);
stream->jitterbuffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL);
if (!stream->jitterbuffer)
goto no_jitterbuffer;
stream->ptdemux = gst_element_factory_make ("rtpptdemux", NULL);
stream->ptdemux = gst_element_factory_make ("gstrtpptdemux", NULL);
if (!stream->ptdemux)
goto no_ptdemux;
@ -180,14 +180,14 @@ create_stream (GstRTPClient * rtpclient, guint32 ssrc)
no_jitterbuffer:
{
g_free (stream);
g_warning ("could not create rtpjitterbuffer element");
g_warning ("gstrtpclient: could not create gstrtpjitterbuffer element");
return NULL;
}
no_ptdemux:
{
gst_object_unref (stream->jitterbuffer);
g_free (stream);
g_warning ("could not create rtpptdemux element");
g_warning ("gstrtpclient: could not create gstrtpptdemux element");
return NULL;
}
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->ptdemux);
g_free (stream);
g_warning ("could not link jitterbuffer and rtpptdemux element");
g_warning ("gstrtpclient: could not link jitterbuffer and ptdemux element");
return NULL;
}
}
@ -455,27 +455,27 @@ gst_rtp_client_request_new_pad (GstElement * element,
/* ERRORS */
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;
}
wrong_template:
{
g_warning ("rtpclient: this is not our template");
g_warning ("gstrtpclient: this is not our template");
return NULL;
}
no_name:
{
g_warning ("rtpclient: no padname was specified");
g_warning ("gstrtpclient: no padname was specified");
return NULL;
}
stream_exists:
{
g_warning ("rtpclient: stream with SSRC %d already registered", ssrc);
g_warning ("gstrtpclient: stream with SSRC %d already registered", ssrc);
return NULL;
}
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;
}
}

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
* compensate for network oddities.
*
@ -33,7 +33,7 @@
* 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
* 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>
* 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.
* </para>
* <para>
* This element will automatically be used inside rtpbin.
* This element will automatically be used inside gstrtpbin.
* </para>
* <title>Example pipelines</title>
* <para>
* <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>
* 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
@ -58,7 +58,7 @@
* </para>
* </refsect2>
*
* Last reviewed on 2007-05-22 (0.10.6)
* Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H

15
gst/rtpmanager/gstrtpmanager.c
View file

@ -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)

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
*
* <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)
*/
/*

34
gst/rtpmanager/gstrtpsession.c
View file

@ -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;
}
}

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
*
* <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