*/
/**
- * 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>
* 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
* <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
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);
/* 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;
}
}
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);
/* 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;
}
}
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("gstrtpbin: invalid name given");
return NULL;
}
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;
}
}
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("gstrtpbin: invalid name given");
return NULL;
}
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
/* ERRORS */
no_name:
{
- g_warning ("rtpbin: invalid name given");
+ g_warning ("gstrtpbin: invalid name given");
return NULL;
}
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;
}
}
/* 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;
}
}
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;
}
}
*/
/**
- * 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>
* </para>
* </refsect2>
*
- * Last reviewed on 2007-04-02 (0.10.6)
+ * Last reviewed on 2007-04-02 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
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;
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:
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;
}
}
/* 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;
}
}
*/
/**
- * SECTION:element-rtpjitterbuffer
+ * SECTION:element-gstrtpjitterbuffer
* @short_description: buffer, reorder and remove duplicate RTP packets to
* compensate for network oddities.
*
* 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.
* 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
* </para>
* </refsect2>
*
- * Last reviewed on 2007-05-22 (0.10.6)
+ * Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
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;
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)
*/
/**
- * 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>
* <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)
*/
/*
*/
/**
- * 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>
* </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
* 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.
* <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
* </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>
* </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
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;
}
}
*/
/**
- * 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>
* <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
projects / platform / upstream / gst-plugins-good.git / commitdiff