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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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a6aa41dc21
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init), (gst_rtp_bin_init), (gst_rtp_bin_provide_clock): * gst/rtpmanager/gstrtpbin.h: Provide a clock.
894 lines
23 KiB
C
894 lines
23 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-rtpbin
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* @short_description: handle media from one RTP bin
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* @see_also: rtpjitterbuffer, rtpclient, rtpsession
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*
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* <refsect2>
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* <para>
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* </para>
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* <title>Example pipelines</title>
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* <para>
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* <programlisting>
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* gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
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* </programlisting>
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-04-02 (0.10.6)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <string.h>
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#include "gstrtpbin.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
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#define GST_CAT_DEFAULT gst_rtp_bin_debug
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/* elementfactory information */
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static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin",
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"Filter/Editor/Video",
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"Implement an RTP bin",
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"Wim Taymans <wim@fluendo.com>");
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/* sink pads */
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static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%d",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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/* src pads */
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static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtpbin_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_src_%d",
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GST_PAD_SRC,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtpbin_send_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%d",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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#define GST_RTP_BIN_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRTPBinPrivate))
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struct _GstRTPBinPrivate
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{
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guint foo;
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};
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/* signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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/* helper objects */
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typedef struct _GstRTPBinSession GstRTPBinSession;
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typedef struct _GstRTPBinStream GstRTPBinStream;
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typedef struct _GstRTPBinClient GstRTPBinClient;
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/* Manages the RTP stream for one SSRC.
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*
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* We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
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* If we see an SDES RTCP packet that links multiple SSRCs together based on a
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* common CNAME, we create a GstRTPBinClient structure to group the SSRCs
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* together (see below).
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*/
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struct _GstRTPBinStream
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{
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/* the SSRC of this stream */
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guint32 ssrc;
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/* parent bin */
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GstRTPBin *bin;
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/* the session this SSRC belongs to */
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GstRTPBinSession *session;
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/* the jitterbuffer of the SSRC */
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GstElement *buffer;
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/* the PT demuxer of the SSRC */
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GstElement *demux;
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gulong demux_newpad_sig;
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};
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/* Manages the receiving end of the packets.
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*
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* There is one such structure for each RTP session (audio/video/...).
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* We get the RTP/RTCP packets and stuff them into the session manager. From
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* there they are pushed into an SSRC demuxer that splits the stream based on
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* SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
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* the GstRTPBinStream above).
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*/
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struct _GstRTPBinSession
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{
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/* session id */
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gint id;
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/* the parent bin */
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GstRTPBin *bin;
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/* the session element */
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GstElement *session;
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/* the SSRC demuxer */
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GstElement *demux;
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gulong demux_newpad_sig;
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/* list of GstRTPBinStream */
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GSList *streams;
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/* the pads of the session */
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GstPad *recv_rtp_sink;
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GstPad *recv_rtp_src;
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GstPad *recv_rtcp_sink;
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GstPad *recv_rtcp_src;
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GstPad *send_rtp_sink;
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GstPad *send_rtp_src;
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GstPad *rtcp_src;
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};
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/* find a session with the given id */
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static GstRTPBinSession *
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find_session_by_id (GstRTPBin * rtpbin, gint id)
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{
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GSList *walk;
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for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
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GstRTPBinSession *sess = (GstRTPBinSession *) walk->data;
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if (sess->id == id)
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return sess;
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}
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return NULL;
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}
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/* create a session with the given id */
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static GstRTPBinSession *
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create_session (GstRTPBin * rtpbin, gint id)
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{
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GstRTPBinSession *sess;
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GstElement *elem, *demux;
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if (!(elem = gst_element_factory_make ("rtpsession", NULL)))
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goto no_session;
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if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
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goto no_demux;
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sess = g_new0 (GstRTPBinSession, 1);
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sess->id = id;
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sess->bin = rtpbin;
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sess->session = elem;
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sess->demux = demux;
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rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
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gst_bin_add (GST_BIN_CAST (rtpbin), elem);
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gst_element_set_state (elem, GST_STATE_PLAYING);
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gst_bin_add (GST_BIN_CAST (rtpbin), demux);
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gst_element_set_state (demux, GST_STATE_PLAYING);
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return sess;
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/* ERRORS */
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no_session:
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{
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g_warning ("rtpbin: could not create rtpsession element");
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return NULL;
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}
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no_demux:
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{
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gst_object_unref (elem);
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g_warning ("rtpbin: could not create rtpssrcdemux element");
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return NULL;
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}
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}
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#if 0
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static GstRTPBinStream *
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find_stream_by_ssrc (GstRTPBinSession * session, guint32 ssrc)
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{
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GSList *walk;
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for (walk = session->streams; walk; walk = g_slist_next (walk)) {
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GstRTPBinStream *stream = (GstRTPBinStream *) walk->data;
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if (stream->ssrc == ssrc)
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return stream;
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}
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return NULL;
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}
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#endif
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static GstRTPBinStream *
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create_stream (GstRTPBinSession * session, guint32 ssrc)
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{
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GstElement *buffer, *demux;
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GstRTPBinStream *stream;
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if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
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goto no_jitterbuffer;
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if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
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goto no_demux;
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stream = g_new0 (GstRTPBinStream, 1);
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stream->ssrc = ssrc;
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stream->bin = session->bin;
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stream->session = session;
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stream->buffer = buffer;
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stream->demux = demux;
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session->streams = g_slist_prepend (session->streams, stream);
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gst_bin_add (GST_BIN_CAST (session->bin), buffer);
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gst_element_set_state (buffer, GST_STATE_PLAYING);
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gst_bin_add (GST_BIN_CAST (session->bin), demux);
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gst_element_set_state (demux, GST_STATE_PLAYING);
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/* link stuff */
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gst_element_link (buffer, demux);
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return stream;
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/* ERRORS */
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no_jitterbuffer:
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{
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g_warning ("rtpbin: could not create rtpjitterbuffer element");
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return NULL;
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}
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no_demux:
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{
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gst_object_unref (buffer);
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g_warning ("rtpbin: could not create rtpptdemux element");
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return NULL;
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}
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}
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/* Manages the RTP streams that come from one client and should therefore be
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* synchronized.
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*/
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struct _GstRTPBinClient
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{
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/* the common CNAME for the streams */
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gchar *cname;
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/* the streams */
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GSList *streams;
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};
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/* GObject vmethods */
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static void gst_rtp_bin_finalize (GObject * object);
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static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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/* GstElement vmethods */
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static GstClock *gst_rtp_bin_provide_clock (GstElement * element);
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static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
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GstStateChange transition);
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static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name);
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static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
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/*static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 }; */
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GST_BOILERPLATE (GstRTPBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
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static void
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gst_rtp_bin_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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/* sink pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
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/* src pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_rtcp_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
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gst_element_class_set_details (element_class, &rtpbin_details);
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}
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static void
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gst_rtp_bin_class_init (GstRTPBinClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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g_type_class_add_private (klass, sizeof (GstRTPBinPrivate));
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gobject_class->finalize = gst_rtp_bin_finalize;
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gobject_class->set_property = gst_rtp_bin_set_property;
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gobject_class->get_property = gst_rtp_bin_get_property;
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gstelement_class->provide_clock =
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GST_DEBUG_FUNCPTR (gst_rtp_bin_provide_clock);
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
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gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
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}
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static void
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gst_rtp_bin_init (GstRTPBin * rtpbin, GstRTPBinClass * klass)
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{
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rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
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rtpbin->provided_clock = gst_system_clock_obtain ();
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}
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static void
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gst_rtp_bin_finalize (GObject * object)
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{
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GstRTPBin *rtpbin;
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rtpbin = GST_RTP_BIN (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_bin_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRTPBin *rtpbin;
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rtpbin = GST_RTP_BIN (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_bin_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRTPBin *rtpbin;
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rtpbin = GST_RTP_BIN (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstClock *
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gst_rtp_bin_provide_clock (GstElement * element)
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{
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GstRTPBin *rtpbin;
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rtpbin = GST_RTP_BIN (element);
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return GST_CLOCK_CAST (gst_object_ref (rtpbin->provided_clock));
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}
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static GstStateChangeReturn
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gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn res;
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GstRTPBin *rtpbin;
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rtpbin = GST_RTP_BIN (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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break;
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default:
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break;
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}
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res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return res;
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}
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/* a new pad (SSRC) was created in @session */
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static void
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new_payload_found (GstElement * element, guint pt, GstPad * pad,
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GstRTPBinStream * stream)
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{
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GstRTPBin *rtpbin;
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GstElementClass *klass;
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GstPadTemplate *templ;
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gchar *padname;
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GstPad *gpad;
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rtpbin = stream->bin;
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GST_DEBUG ("new payload pad %d", pt);
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/* ghost the pad to the parent */
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klass = GST_ELEMENT_GET_CLASS (rtpbin);
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templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
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padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
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stream->session->id, stream->ssrc, pt);
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gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
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g_free (padname);
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gst_pad_set_active (gpad, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
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}
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/* a new pad (SSRC) was created in @session */
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|
static void
|
|
new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
|
|
GstRTPBinSession * session)
|
|
{
|
|
GstRTPBinStream *stream;
|
|
GstPad *sinkpad;
|
|
|
|
GST_DEBUG_OBJECT (session->bin, "new SSRC pad %08x", ssrc);
|
|
|
|
/* create new stream */
|
|
stream = create_stream (session, ssrc);
|
|
if (!stream)
|
|
goto no_stream;
|
|
|
|
/* get pad and link */
|
|
GST_DEBUG_OBJECT (session->bin, "linking jitterbuffer");
|
|
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
|
|
gst_pad_link (pad, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
|
|
/* connect to the new-pad signal of the payload demuxer */
|
|
stream->demux_newpad_sig = g_signal_connect (stream->demux,
|
|
"new-payload-type", (GCallback) new_payload_found, stream);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_stream:
|
|
{
|
|
GST_DEBUG ("could not create stream");
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for receiving RTP for the session in @name
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstPad *result, *sinkdpad;
|
|
guint sessid;
|
|
GstRTPBinSession *session;
|
|
GstPadLinkReturn lres;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
|
|
/* create session now */
|
|
session = create_session (rtpbin, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
/* check if pad was requested */
|
|
if (session->recv_rtp_sink != NULL)
|
|
goto existed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
|
|
/* get recv_rtp pad and store */
|
|
session->recv_rtp_sink =
|
|
gst_element_get_request_pad (session->session, "recv_rtp_sink");
|
|
if (session->recv_rtp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
|
|
/* get srcpad, link to SSRCDemux */
|
|
session->recv_rtp_src =
|
|
gst_element_get_static_pad (session->session, "recv_rtp_src");
|
|
if (session->recv_rtp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting demuxer sink pad");
|
|
sinkdpad = gst_element_get_static_pad (session->demux, "sink");
|
|
lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
|
|
gst_object_unref (sinkdpad);
|
|
if (lres != GST_PAD_LINK_OK)
|
|
goto link_failed;
|
|
|
|
/* connect to the new-ssrc-pad signal of the demuxer */
|
|
session->demux_newpad_sig = g_signal_connect (session->demux,
|
|
"new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
|
|
result =
|
|
gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
|
|
gst_pad_set_active (result, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rtpbin: recv_rtp pad already requested for session %d", sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session pad");
|
|
return NULL;
|
|
}
|
|
link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link pads");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for receiving RTCP for the session in @name
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ,
|
|
const gchar * name)
|
|
{
|
|
GstPad *result;
|
|
guint sessid;
|
|
GstRTPBinSession *session;
|
|
|
|
#if 0
|
|
GstPad *sinkdpad;
|
|
GstPadLinkReturn lres;
|
|
#endif
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
|
|
|
|
/* get the session, it must exist or we error */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session)
|
|
goto no_session;
|
|
|
|
/* check if pad was requested */
|
|
if (session->recv_rtcp_sink != NULL)
|
|
goto existed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
|
|
|
|
/* get recv_rtp pad and store */
|
|
session->recv_rtcp_sink =
|
|
gst_element_get_request_pad (session->session, "recv_rtcp_sink");
|
|
if (session->recv_rtcp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
#if 0
|
|
/* get srcpad, link to SSRCDemux */
|
|
GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
|
|
session->recv_rtcp_src =
|
|
gst_element_get_static_pad (session->session, "sync_src");
|
|
if (session->recv_rtcp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "linking sync to demux");
|
|
sinkdpad = gst_element_get_static_pad (session->demux, "sink");
|
|
lres = gst_pad_link (session->recv_rtcp_src, sinkdpad);
|
|
gst_object_unref (sinkdpad);
|
|
if (lres != GST_PAD_LINK_OK)
|
|
goto link_failed;
|
|
#endif
|
|
|
|
result =
|
|
gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
|
|
gst_pad_set_active (result, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("rtpbin: no session with id %d", sessid);
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rtpbin: recv_rtcp pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session pad");
|
|
return NULL;
|
|
}
|
|
#if 0
|
|
link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link pads");
|
|
return NULL;
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/* Create a pad for sending RTP for the session in @name
|
|
*/
|
|
static GstPad *
|
|
create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstPad *result, *srcpad, *srcghost;
|
|
gchar *gname;
|
|
guint sessid;
|
|
GstRTPBinSession *session;
|
|
GstElementClass *klass;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "send_rtp_sink_%d", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
/* create session now */
|
|
session = create_session (rtpbin, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
|
|
/* check if pad was requested */
|
|
if (session->send_rtp_sink != NULL)
|
|
goto existed;
|
|
|
|
/* get recv_rtp pad and store */
|
|
session->send_rtp_sink =
|
|
gst_element_get_request_pad (session->session, "send_rtp_sink");
|
|
if (session->send_rtp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
result =
|
|
gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
|
|
gst_pad_set_active (result, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
|
|
|
|
/* get srcpad */
|
|
srcpad = gst_element_get_pad (session->session, "send_rtp_src");
|
|
if (srcpad == NULL)
|
|
goto no_srcpad;
|
|
|
|
/* ghost the new source pad */
|
|
klass = GST_ELEMENT_GET_CLASS (rtpbin);
|
|
gname = g_strdup_printf ("send_rtp_src_%d", sessid);
|
|
templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
|
|
srcghost =
|
|
gst_ghost_pad_new_from_template (gname, session->send_rtp_sink, templ);
|
|
gst_pad_set_active (srcghost, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), srcghost);
|
|
g_free (gname);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rtpbin: 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);
|
|
return NULL;
|
|
}
|
|
no_srcpad:
|
|
{
|
|
g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for sending RTCP for the session in @name
|
|
*/
|
|
static GstPad *
|
|
create_rtcp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstPad *result;
|
|
guint sessid;
|
|
GstRTPBinSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "rtcp_src_%d", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session)
|
|
goto no_session;
|
|
|
|
/* check if pad was requested */
|
|
if (session->rtcp_src != NULL)
|
|
goto existed;
|
|
|
|
/* get rtcp_src pad and store */
|
|
session->rtcp_src =
|
|
gst_element_get_request_pad (session->session, "rtcp_src");
|
|
if (session->rtcp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
result = gst_ghost_pad_new_from_template (name, session->rtcp_src, templ);
|
|
gst_pad_set_active (result, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("rtpbin: session with id %d does not exist", sessid);
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("rtpbin: 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);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/*
|
|
*/
|
|
static GstPad *
|
|
gst_rtp_bin_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstRTPBin *rtpbin;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
|
|
|
|
rtpbin = GST_RTP_BIN (element);
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
|
|
result = create_recv_rtp (rtpbin, templ, name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"recv_rtcp_sink_%d")) {
|
|
result = create_recv_rtcp (rtpbin, templ, name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"send_rtp_sink_%d")) {
|
|
result = create_send_rtp (rtpbin, templ, name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%d")) {
|
|
result = create_rtcp (rtpbin, templ, name);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_warning ("rtpbin: this is not our template");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
}
|