diff options
Diffstat (limited to 'gst/rtpmanager')
| -rw-r--r-- | gst/rtpmanager/gstrtpbin.c | 89 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpclient.c | 26 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpjitterbuffer.c | 10 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpmanager.c | 15 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpptdemux.c | 8 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpsession.c | 34 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpssrcdemux.c | 8 |
7 files changed, 97 insertions, 93 deletions
diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 44fe3386..cc0afd00 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -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; } } diff --git a/gst/rtpmanager/gstrtpclient.c b/gst/rtpmanager/gstrtpclient.c index 86c5f3cc..ba18f953 100644 --- a/gst/rtpmanager/gstrtpclient.c +++ b/gst/rtpmanager/gstrtpclient.c @@ -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; } } diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 1838ba0b..33f78741 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -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 diff --git a/gst/rtpmanager/gstrtpmanager.c b/gst/rtpmanager/gstrtpmanager.c index 1490c4cb..eed5c25f 100644 --- a/gst/rtpmanager/gstrtpmanager.c +++ b/gst/rtpmanager/gstrtpmanager.c @@ -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) diff --git a/gst/rtpmanager/gstrtpptdemux.c b/gst/rtpmanager/gstrtpptdemux.c index 578f0353..235bbdd4 100644 --- a/gst/rtpmanager/gstrtpptdemux.c +++ b/gst/rtpmanager/gstrtpptdemux.c @@ -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) */ /* diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 431098d9..3e33cf6a 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -18,9 +18,9 @@ */ /** - * SECTION:element-rtpsession + * SECTION:element-gstrtpsession * @short_description: an RTP session manager - * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux + * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux * * <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; } } diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c index bd0c6b05..e1100be4 100644 --- a/gst/rtpmanager/gstrtpssrcdemux.c +++ b/gst/rtpmanager/gstrtpssrcdemux.c @@ -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 |