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9e50d836d4
Original commit message from CVS: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/async_jitter_queue.c: * gst/rtpmanager/async_jitter_queue.h: * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init), (rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize), (rtp_jitter_buffer_new), (compare_seqnum), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop), (rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets), (rtp_jitter_buffer_get_ts_diff): * gst/rtpmanager/rtpjitterbuffer.h: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some bugs. * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (create_session), (gst_rtp_bin_class_init), (create_recv_rtp), (create_send_rtp): * gst/rtpmanager/gstrtpbin.h: * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose), (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_flush_start), (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property): * gst/rtpmanager/gstrtpsession.c: (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: * gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup): Use new jitterbuffer code. Expose some new signals in preparation for handling EOS.
1299 lines
36 KiB
C
1299 lines
36 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-gstrtpbin
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* @short_description: handle media from one RTP bin
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* @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
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*
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* <refsect2>
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* <para>
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* RTP bin combines the functions of gstrtpsession, gstrtpssrcdemux, gstrtpjitterbuffer
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* and gstrtpptdemux in one element. It allows for multiple RTP sessions that will
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* be synchronized together using RTCP SR packets.
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* </para>
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* <para>
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* gstrtpbin is configured with a number of request pads that define the
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* functionality that is activated, similar to the gstrtpsession element.
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* </para>
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* <para>
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* To use gstrtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
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* number must be specified in the pad name.
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* Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
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* manager and after being validated forwarded on gstrtpssrcdemuxer element. Each
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* RTP stream is demuxed based on the SSRC and send to a gstrtpjitterbuffer. After
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* the packets are released from the jitterbuffer, they will be forwarded to a
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* gstrtpptdemuxer element. The gstrtpptdemuxer element will demux the packets based
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* on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
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* gstrtpbin with the session number, SSRC and payload type respectively as the pad
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* name.
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* </para>
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* <para>
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* To also use gstrtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
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* session number must be specified in the pad name.
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* </para>
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* <para>
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* If you want the session manager to generate and send RTCP packets, request
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* the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed
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* on this pad contain SR/RR RTCP reports that should be sent to all participants
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* in the session.
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* </para>
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* <para>
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* To use gstrtpbin as a sender, request a send_rtp_sink_%%d pad, which will
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* automatically create a send_rtp_src_%%d pad. The session number must be specified when
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* requesting the sink pad. The session manager will modify the
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* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
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* send_rtp_src_%%d pad after updating its internal state.
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* </para>
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* <para>
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* The session manager needs the clock-rate of the payload types it is handling
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* and will signal the GstRTPSession::request-pt-map signal when it needs such a
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* mapping. One can clear the cached values with the GstRTPSession::clear-pt-map
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* signal.
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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 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
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* gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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* </programlisting>
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* Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
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* </para>
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* </refsect2>
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*
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* Last reviewed on 2007-05-28 (0.10.5)
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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-marshal.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/Network/RTP",
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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_send_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_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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#define GST_RTP_BIN_LOCK(bin) g_mutex_lock ((bin)->priv->bin_lock)
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#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->bin_lock)
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struct _GstRTPBinPrivate
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{
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GMutex *bin_lock;
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};
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/* signals and args */
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enum
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{
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SIGNAL_REQUEST_PT_MAP,
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SIGNAL_CLEAR_PT_MAP,
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SIGNAL_ON_NEW_SSRC,
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SIGNAL_ON_SSRC_COLLISION,
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SIGNAL_ON_SSRC_VALIDATED,
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SIGNAL_ON_BYE_SSRC,
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SIGNAL_ON_BYE_TIMEOUT,
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SIGNAL_ON_TIMEOUT,
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LAST_SIGNAL
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};
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#define DEFAULT_LATENCY_MS 200
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enum
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{
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PROP_0,
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PROP_LATENCY
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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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static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
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static GstCaps *pt_map_requested (GstElement * element, guint pt,
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GstRTPBinSession * session);
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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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gulong demux_ptreq_sig;
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};
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#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
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#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->lock)
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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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GMutex *lock;
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/* list of GstRTPBinStream */
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GSList *streams;
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/* mapping of payload type to caps */
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GHashTable *ptmap;
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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 *send_rtcp_src;
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};
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/* find a session with the given id. Must be called with RTP_BIN_LOCK */
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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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static void
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on_new_ssrc (GstElement * session, guint32 ssrc, GstRTPBinSession * sess)
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{
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g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
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sess->id, ssrc);
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}
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static void
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on_ssrc_collision (GstElement * session, guint32 ssrc, GstRTPBinSession * sess)
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{
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g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
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sess->id, ssrc);
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}
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static void
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on_ssrc_validated (GstElement * session, guint32 ssrc, GstRTPBinSession * sess)
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{
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g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
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sess->id, ssrc);
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}
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static void
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on_bye_ssrc (GstElement * session, guint32 ssrc, GstRTPBinSession * sess)
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{
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g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
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sess->id, ssrc);
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}
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static void
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on_bye_timeout (GstElement * session, guint32 ssrc, GstRTPBinSession * sess)
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{
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g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
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sess->id, ssrc);
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}
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static void
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on_timeout (GstElement * session, guint32 ssrc, GstRTPBinSession * sess)
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{
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g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
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sess->id, ssrc);
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}
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/* create a session with the given id. Must be called with RTP_BIN_LOCK */
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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 *session, *demux;
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if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
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goto no_session;
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if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
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goto no_demux;
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sess = g_new0 (GstRTPBinSession, 1);
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sess->lock = g_mutex_new ();
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sess->id = id;
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sess->bin = rtpbin;
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sess->session = session;
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sess->demux = demux;
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sess->ptmap = g_hash_table_new (NULL, NULL);
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rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
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/* provide clock_rate to the session manager when needed */
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g_signal_connect (session, "request-pt-map",
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(GCallback) pt_map_requested, sess);
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g_signal_connect (sess->session, "on-new-ssrc",
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(GCallback) on_new_ssrc, sess);
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g_signal_connect (sess->session, "on-ssrc-collision",
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(GCallback) on_ssrc_collision, sess);
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g_signal_connect (sess->session, "on-ssrc-validated",
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(GCallback) on_ssrc_validated, sess);
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g_signal_connect (sess->session, "on-bye-ssrc",
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(GCallback) on_bye_ssrc, sess);
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g_signal_connect (sess->session, "on-bye-timeout",
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(GCallback) on_bye_timeout, sess);
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g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
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gst_bin_add (GST_BIN_CAST (rtpbin), session);
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gst_element_set_state (session, 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 ("gstrtpbin: could not create gstrtpsession 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 (session);
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g_warning ("gstrtpbin: could not create gstrtpssrcdemux 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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|
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/* get the payload type caps for the specific payload @pt in @session */
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static GstCaps *
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get_pt_map (GstRTPBinSession * session, guint pt)
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{
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GstCaps *caps = NULL;
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GstRTPBin *bin;
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GValue ret = { 0 };
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GValue args[3] = { {0}, {0}, {0} };
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GST_DEBUG ("searching pt %d in cache", pt);
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GST_RTP_SESSION_LOCK (session);
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/* first look in the cache */
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caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
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if (caps)
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goto done;
|
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|
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bin = session->bin;
|
|
|
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GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
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|
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/* not in cache, send signal to request caps */
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g_value_init (&args[0], GST_TYPE_ELEMENT);
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g_value_set_object (&args[0], bin);
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g_value_init (&args[1], G_TYPE_UINT);
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g_value_set_uint (&args[1], session->id);
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g_value_init (&args[2], G_TYPE_UINT);
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g_value_set_uint (&args[2], pt);
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|
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g_value_init (&ret, GST_TYPE_CAPS);
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g_value_set_boxed (&ret, NULL);
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|
|
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g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
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|
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caps = (GstCaps *) g_value_get_boxed (&ret);
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if (!caps)
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goto no_caps;
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|
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GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
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|
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/* store in cache */
|
|
g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt), caps);
|
|
|
|
done:
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
|
|
return caps;
|
|
|
|
/* ERRORS */
|
|
no_caps:
|
|
{
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
GST_DEBUG ("no pt map could be obtained");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
return_true (gpointer key, gpointer value, gpointer user_data)
|
|
{
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_clear_pt_map (GstRTPBin * bin)
|
|
{
|
|
GSList *walk;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
|
|
GstRTPBinSession *session = (GstRTPBinSession *) walk->data;
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
#if 0
|
|
/* This requires GLib 2.12 */
|
|
g_hash_table_remove_all (session->ptmap);
|
|
#else
|
|
g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
|
|
#endif
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
|
|
/* create a new stream with @ssrc in @session. Must be called with
|
|
* RTP_SESSION_LOCK. */
|
|
static GstRTPBinStream *
|
|
create_stream (GstRTPBinSession * session, guint32 ssrc)
|
|
{
|
|
GstElement *buffer, *demux;
|
|
GstRTPBinStream *stream;
|
|
|
|
if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
|
|
goto no_jitterbuffer;
|
|
|
|
if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
|
|
goto no_demux;
|
|
|
|
stream = g_new0 (GstRTPBinStream, 1);
|
|
stream->ssrc = ssrc;
|
|
stream->bin = session->bin;
|
|
stream->session = session;
|
|
stream->buffer = buffer;
|
|
stream->demux = demux;
|
|
session->streams = g_slist_prepend (session->streams, stream);
|
|
|
|
/* provide clock_rate to the jitterbuffer when needed */
|
|
g_signal_connect (buffer, "request-pt-map",
|
|
(GCallback) pt_map_requested, session);
|
|
|
|
/* configure latency */
|
|
g_object_set (buffer, "latency", session->bin->latency, NULL);
|
|
|
|
gst_bin_add (GST_BIN_CAST (session->bin), buffer);
|
|
gst_element_set_state (buffer, GST_STATE_PLAYING);
|
|
gst_bin_add (GST_BIN_CAST (session->bin), demux);
|
|
gst_element_set_state (demux, GST_STATE_PLAYING);
|
|
|
|
/* link stuff */
|
|
gst_element_link (buffer, demux);
|
|
|
|
return stream;
|
|
|
|
/* ERRORS */
|
|
no_jitterbuffer:
|
|
{
|
|
g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
|
|
return NULL;
|
|
}
|
|
no_demux:
|
|
{
|
|
gst_object_unref (buffer);
|
|
g_warning ("gstrtpbin: could not create gstrtpptdemux element");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Manages the RTP streams that come from one client and should therefore be
|
|
* synchronized.
|
|
*/
|
|
struct _GstRTPBinClient
|
|
{
|
|
/* the common CNAME for the streams */
|
|
gchar *cname;
|
|
/* the streams */
|
|
GSList *streams;
|
|
};
|
|
|
|
/* GObject vmethods */
|
|
static void gst_rtp_bin_finalize (GObject * object);
|
|
static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
|
|
/* GstElement vmethods */
|
|
static GstClock *gst_rtp_bin_provide_clock (GstElement * element);
|
|
static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
|
|
GstStateChange transition);
|
|
static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name);
|
|
static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
|
|
static void gst_rtp_bin_clear_pt_map (GstRTPBin * bin);
|
|
|
|
GST_BOILERPLATE (GstRTPBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
|
|
|
|
static void
|
|
gst_rtp_bin_base_init (gpointer klass)
|
|
{
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
|
|
/* sink pads */
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
|
|
|
|
/* src pads */
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
|
|
|
|
gst_element_class_set_details (element_class, &rtpbin_details);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_class_init (GstRTPBinClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
|
|
g_type_class_add_private (klass, sizeof (GstRTPBinPrivate));
|
|
|
|
gobject_class->finalize = gst_rtp_bin_finalize;
|
|
gobject_class->set_property = gst_rtp_bin_set_property;
|
|
gobject_class->get_property = gst_rtp_bin_get_property;
|
|
|
|
g_object_class_install_property (gobject_class, PROP_LATENCY,
|
|
g_param_spec_uint ("latency", "Buffer latency in ms",
|
|
"Default amount of ms to buffer in the jitterbuffers", 0,
|
|
G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE));
|
|
|
|
/**
|
|
* GstRTPBin::request-pt-map:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @pt: the pt
|
|
*
|
|
* Request the payload type as #GstCaps for @pt in @session.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
|
|
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, request_pt_map),
|
|
NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRTPBin::clear-pt-map:
|
|
* @rtpbin: the object which received the signal
|
|
*
|
|
* Clear all previously cached pt-mapping obtained with
|
|
* GstRTPBin::request-pt-map.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
|
|
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTPBinClass, clear_pt_map),
|
|
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* GstRTPBin::on-new-ssrc:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a new SSRC that entered @session.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
|
|
g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, on_new_ssrc),
|
|
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRTPBin::on-ssrc_collision:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify when we have an SSRC collision
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
|
|
g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, on_ssrc_collision),
|
|
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRTPBin::on-ssrc_validated:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a new SSRC that became validated.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
|
|
g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, on_ssrc_validated),
|
|
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRTPBin::on-bye-ssrc:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that became inactive because of a BYE packet.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
|
|
g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, on_bye_ssrc),
|
|
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRTPBin::on-bye-timeout:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that has timed out because of BYE
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
|
|
g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, on_bye_timeout),
|
|
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRTPBin::on-timeout:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that has timed out
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
|
|
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPBinClass, on_timeout),
|
|
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
gstelement_class->provide_clock =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_bin_provide_clock);
|
|
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
|
|
gstelement_class->request_new_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
|
|
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
|
|
|
|
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_init (GstRTPBin * rtpbin, GstRTPBinClass * klass)
|
|
{
|
|
rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
|
|
rtpbin->priv->bin_lock = g_mutex_new ();
|
|
rtpbin->provided_clock = gst_system_clock_obtain ();
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_finalize (GObject * object)
|
|
{
|
|
GstRTPBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (object);
|
|
|
|
g_mutex_free (rtpbin->priv->bin_lock);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTPBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
rtpbin->latency = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRTPBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
g_value_set_uint (value, rtpbin->latency);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstClock *
|
|
gst_rtp_bin_provide_clock (GstElement * element)
|
|
{
|
|
GstRTPBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (element);
|
|
|
|
return GST_CLOCK_CAST (gst_object_ref (rtpbin->provided_clock));
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn res;
|
|
GstRTPBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
/* a new pad (SSRC) was created in @session */
|
|
static void
|
|
new_payload_found (GstElement * element, guint pt, GstPad * pad,
|
|
GstRTPBinStream * stream)
|
|
{
|
|
GstRTPBin *rtpbin;
|
|
GstElementClass *klass;
|
|
GstPadTemplate *templ;
|
|
gchar *padname;
|
|
GstPad *gpad;
|
|
|
|
rtpbin = stream->bin;
|
|
|
|
GST_DEBUG ("new payload pad %d", pt);
|
|
|
|
/* ghost the pad to the parent */
|
|
klass = GST_ELEMENT_GET_CLASS (rtpbin);
|
|
templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
|
|
padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
|
|
stream->session->id, stream->ssrc, pt);
|
|
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
|
|
g_free (padname);
|
|
|
|
gst_pad_set_active (gpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
|
|
}
|
|
|
|
static GstCaps *
|
|
pt_map_requested (GstElement * element, guint pt, GstRTPBinSession * session)
|
|
{
|
|
GstRTPBin *rtpbin;
|
|
GstCaps *caps;
|
|
|
|
rtpbin = session->bin;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
|
|
session->id);
|
|
|
|
caps = get_pt_map (session, pt);
|
|
if (!caps)
|
|
goto no_caps;
|
|
|
|
return caps;
|
|
|
|
/* ERRORS */
|
|
no_caps:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpbin, "could not get caps");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* a new pad (SSRC) was created in @session */
|
|
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);
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
|
|
/* 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, this will expose the
|
|
* new pad by ghosting it. */
|
|
stream->demux_newpad_sig = g_signal_connect (stream->demux,
|
|
"new-payload-type", (GCallback) new_payload_found, stream);
|
|
/* connect to the request-pt-map signal. This signal will be emited by the
|
|
* demuxer so that it can apply a proper caps on the buffers for the
|
|
* depayloaders. */
|
|
stream->demux_ptreq_sig = g_signal_connect (stream->demux,
|
|
"request-pt-map", (GCallback) pt_map_requested, session);
|
|
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_stream:
|
|
{
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
GST_DEBUG ("could not create stream");
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for receiving RTP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
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 SSRC 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 ("gstrtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("gstrtpbin: recv_rtp pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("gstrtpbin: failed to get session pad");
|
|
return NULL;
|
|
}
|
|
link_failed:
|
|
{
|
|
g_warning ("gstrtpbin: failed to link pads");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for receiving RTCP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
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 or create the 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_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 ("gstrtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("gstrtpbin: failed to get session pad");
|
|
return NULL;
|
|
}
|
|
#if 0
|
|
link_failed:
|
|
{
|
|
g_warning ("gstrtpbin: failed to link pads");
|
|
return NULL;
|
|
}
|
|
#endif
|
|
}
|
|
|
|
/* Create a pad for sending RTP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
static GstPad *
|
|
create_send_rtp (GstRTPBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
GstPad *result, *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 send_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 */
|
|
session->send_rtp_src =
|
|
gst_element_get_static_pad (session->session, "send_rtp_src");
|
|
if (session->send_rtp_src == 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_src, 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 ("gstrtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("gstrtpbin: send_rtp pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
|
|
return NULL;
|
|
}
|
|
no_srcpad:
|
|
{
|
|
g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for sending RTCP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
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, "send_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->send_rtcp_src != NULL)
|
|
goto existed;
|
|
|
|
/* get rtcp_src pad and store */
|
|
session->send_rtcp_src =
|
|
gst_element_get_request_pad (session->session, "send_rtcp_src");
|
|
if (session->send_rtcp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
result =
|
|
gst_ghost_pad_new_from_template (name, session->send_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 ("gstrtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("gstrtpbin: session with id %d does not exist", sessid);
|
|
return NULL;
|
|
}
|
|
existed:
|
|
{
|
|
g_warning ("gstrtpbin: send_rtcp_src pad already requested for session %d",
|
|
sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("gstrtpbin: 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);
|
|
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
|
|
/* 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,
|
|
"send_rtcp_src_%d")) {
|
|
result = create_rtcp (rtpbin, templ, name);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
g_warning ("gstrtpbin: this is not our template");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
}
|