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fdd091e26a
Original commit message from CVS: * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_change_state): Don't clean up pads when going to PAUSED.
655 lines
18 KiB
C
655 lines
18 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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*
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* RTP SSRC demuxer
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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-gstrtpssrcdemux
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* @short_description: separate RTP payloads based on the SSRC
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*
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* <refsect2>
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* <para>
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* gstrtpssrcdemux acts as a demuxer for RTP packets based on the SSRC of the
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* packets. Its main purpose is to allow an application to easily receive and
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* decode an RTP stream with multiple SSRCs.
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* </para>
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* <para>
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* For each SSRC that is detected, a new pad will be created and the
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* ::new-ssrc-pad signal will be emitted.
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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 caps="application/x-rtp" ! gstrtpssrcdemux ! fakesink
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* </programlisting>
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* Takes an RTP stream and send the RTP packets with the first detected SSRC
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* to fakesink, discarding the other SSRCs.
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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 <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include "gstrtpbin-marshal.h"
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#include "gstrtpssrcdemux.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_ssrc_demux_debug);
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#define GST_CAT_DEFAULT gst_rtp_ssrc_demux_debug
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/* generic templates */
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static GstStaticPadTemplate rtp_ssrc_demux_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtp_ssrc_demux_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtp_ssrc_demux_src_template =
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GST_STATIC_PAD_TEMPLATE ("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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static GstStaticPadTemplate rtp_ssrc_demux_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_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstElementDetails gst_rtp_ssrc_demux_details = {
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"RTP SSRC Demux",
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"Demux/Network/RTP",
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"Splits RTP streams based on the SSRC",
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"Wim Taymans <wim.taymans@gmail.com>"
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};
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#define GST_PAD_LOCK(obj) (g_mutex_lock ((obj)->padlock))
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#define GST_PAD_UNLOCK(obj) (g_mutex_unlock ((obj)->padlock))
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/* signals */
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enum
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{
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SIGNAL_NEW_SSRC_PAD,
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LAST_SIGNAL
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};
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GST_BOILERPLATE (GstRtpSsrcDemux, gst_rtp_ssrc_demux, GstElement,
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GST_TYPE_ELEMENT);
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/* GObject vmethods */
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static void gst_rtp_ssrc_demux_dispose (GObject * object);
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static void gst_rtp_ssrc_demux_finalize (GObject * object);
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/* GstElement vmethods */
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static GstStateChangeReturn gst_rtp_ssrc_demux_change_state (GstElement *
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element, GstStateChange transition);
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/* sinkpad stuff */
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static GstFlowReturn gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad,
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GstBuffer * buf);
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static gboolean gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad,
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GstEvent * event);
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/* srcpad stuff */
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static gboolean gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event);
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static GList *gst_rtp_ssrc_demux_internal_links (GstPad * pad);
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static gboolean gst_rtp_ssrc_demux_src_query (GstPad * pad, GstQuery * query);
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static guint gst_rtp_ssrc_demux_signals[LAST_SIGNAL] = { 0 };
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/**
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* Item for storing GstPad <-> SSRC pairs.
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*/
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struct _GstRtpSsrcDemuxPad
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{
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guint32 ssrc;
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GstPad *rtp_pad;
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GstCaps *caps;
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GstPad *rtcp_pad;
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GstClockTime first_ts;
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};
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/* find a src pad for a given SSRC, returns NULL if the SSRC was not found
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*/
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static GstRtpSsrcDemuxPad *
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find_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc)
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{
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GSList *walk;
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for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
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GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
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if (pad->ssrc == ssrc)
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return pad;
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}
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return NULL;
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}
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/* with PAD_LOCK */
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static GstRtpSsrcDemuxPad *
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create_demux_pad_for_ssrc (GstRtpSsrcDemux * demux, guint32 ssrc,
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GstClockTime timestamp)
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{
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GstPad *rtp_pad, *rtcp_pad;
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GstElementClass *klass;
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GstPadTemplate *templ;
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gchar *padname;
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GstRtpSsrcDemuxPad *demuxpad;
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GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
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klass = GST_ELEMENT_GET_CLASS (demux);
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templ = gst_element_class_get_pad_template (klass, "src_%d");
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padname = g_strdup_printf ("src_%d", ssrc);
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rtp_pad = gst_pad_new_from_template (templ, padname);
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g_free (padname);
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templ = gst_element_class_get_pad_template (klass, "rtcp_src_%d");
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padname = g_strdup_printf ("rtcp_src_%d", ssrc);
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rtcp_pad = gst_pad_new_from_template (templ, padname);
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g_free (padname);
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/* we use the first timestamp received to calculate the difference between
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* timestamps on all streams */
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GST_DEBUG_OBJECT (demux, "SSRC %08x, first timestamp %" GST_TIME_FORMAT,
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ssrc, GST_TIME_ARGS (timestamp));
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/* wrap in structure and add to list */
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demuxpad = g_new0 (GstRtpSsrcDemuxPad, 1);
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demuxpad->ssrc = ssrc;
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demuxpad->rtp_pad = rtp_pad;
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demuxpad->rtcp_pad = rtcp_pad;
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demuxpad->first_ts = timestamp;
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GST_DEBUG_OBJECT (demux, "first timestamp %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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gst_pad_set_element_private (rtp_pad, demuxpad);
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gst_pad_set_element_private (rtcp_pad, demuxpad);
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demux->srcpads = g_slist_prepend (demux->srcpads, demuxpad);
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/* copy caps from input */
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gst_pad_set_caps (rtp_pad, GST_PAD_CAPS (demux->rtp_sink));
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gst_pad_use_fixed_caps (rtp_pad);
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gst_pad_set_caps (rtcp_pad, GST_PAD_CAPS (demux->rtcp_sink));
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gst_pad_use_fixed_caps (rtcp_pad);
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gst_pad_set_event_function (rtp_pad, gst_rtp_ssrc_demux_src_event);
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gst_pad_set_query_function (rtp_pad, gst_rtp_ssrc_demux_src_query);
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gst_pad_set_internal_link_function (rtp_pad,
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gst_rtp_ssrc_demux_internal_links);
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gst_pad_set_active (rtp_pad, TRUE);
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gst_pad_set_internal_link_function (rtcp_pad,
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gst_rtp_ssrc_demux_internal_links);
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gst_pad_set_active (rtcp_pad, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), rtp_pad);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), rtcp_pad);
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g_signal_emit (G_OBJECT (demux),
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gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD], 0, ssrc, rtp_pad);
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return demuxpad;
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}
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static void
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gst_rtp_ssrc_demux_base_init (gpointer g_class)
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{
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GstElementClass *gstelement_klass = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_sink_template));
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_sink_template));
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_src_template));
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gst_element_class_add_pad_template (gstelement_klass,
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gst_static_pad_template_get (&rtp_ssrc_demux_rtcp_src_template));
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gst_element_class_set_details (gstelement_klass, &gst_rtp_ssrc_demux_details);
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}
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static void
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gst_rtp_ssrc_demux_class_init (GstRtpSsrcDemuxClass * klass)
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{
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GObjectClass *gobject_klass;
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GstElementClass *gstelement_klass;
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gobject_klass = (GObjectClass *) klass;
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gstelement_klass = (GstElementClass *) klass;
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gobject_klass->dispose = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_dispose);
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gobject_klass->finalize = GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_finalize);
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/**
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* GstRtpSsrcDemux::new-ssrc-pad:
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* @demux: the object which received the signal
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* @ssrc: the SSRC of the pad
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* @pad: the new pad.
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*
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* Emited when a new SSRC pad has been created.
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*/
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gst_rtp_ssrc_demux_signals[SIGNAL_NEW_SSRC_PAD] =
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g_signal_new ("new-ssrc-pad",
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G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
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G_STRUCT_OFFSET (GstRtpSsrcDemuxClass, new_ssrc_pad),
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NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_OBJECT,
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G_TYPE_NONE, 2, G_TYPE_UINT, GST_TYPE_PAD);
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gstelement_klass->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_ssrc_demux_change_state);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_ssrc_demux_debug,
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"rtpssrcdemux", 0, "RTP SSRC demuxer");
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}
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static void
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gst_rtp_ssrc_demux_init (GstRtpSsrcDemux * demux,
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GstRtpSsrcDemuxClass * g_class)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (demux);
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demux->rtp_sink =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"sink"), "sink");
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gst_pad_set_chain_function (demux->rtp_sink, gst_rtp_ssrc_demux_chain);
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gst_pad_set_event_function (demux->rtp_sink, gst_rtp_ssrc_demux_sink_event);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtp_sink);
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demux->rtcp_sink =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"rtcp_sink"), "rtcp_sink");
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gst_pad_set_chain_function (demux->rtcp_sink, gst_rtp_ssrc_demux_rtcp_chain);
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gst_pad_set_event_function (demux->rtcp_sink,
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gst_rtp_ssrc_demux_rtcp_sink_event);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), demux->rtcp_sink);
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demux->padlock = g_mutex_new ();
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gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
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}
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static void
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gst_rtp_ssrc_demux_reset (GstRtpSsrcDemux * demux)
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{
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GSList *walk;
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for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
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GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
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gst_pad_set_active (dpad->rtp_pad, FALSE);
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gst_pad_set_active (dpad->rtcp_pad, FALSE);
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gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtp_pad);
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gst_element_remove_pad (GST_ELEMENT_CAST (demux), dpad->rtcp_pad);
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g_free (dpad);
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}
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g_slist_free (demux->srcpads);
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demux->srcpads = NULL;
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}
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static void
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gst_rtp_ssrc_demux_dispose (GObject * object)
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{
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GstRtpSsrcDemux *demux;
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demux = GST_RTP_SSRC_DEMUX (object);
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gst_rtp_ssrc_demux_reset (demux);
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_rtp_ssrc_demux_finalize (GObject * object)
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{
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GstRtpSsrcDemux *demux;
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demux = GST_RTP_SSRC_DEMUX (object);
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g_mutex_free (demux->padlock);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_ssrc_demux_sink_event (GstPad * pad, GstEvent * event)
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{
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GstRtpSsrcDemux *demux;
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gboolean res = FALSE;
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demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_FLUSH_STOP:
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gst_segment_init (&demux->segment, GST_FORMAT_UNDEFINED);
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case GST_EVENT_NEWSEGMENT:
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default:
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{
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GSList *walk;
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res = TRUE;
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GST_PAD_LOCK (demux);
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for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
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GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
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gst_event_ref (event);
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res &= gst_pad_push_event (pad->rtp_pad, event);
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}
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GST_PAD_UNLOCK (demux);
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gst_event_unref (event);
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break;
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}
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}
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gst_object_unref (demux);
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return res;
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}
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static gboolean
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gst_rtp_ssrc_demux_rtcp_sink_event (GstPad * pad, GstEvent * event)
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{
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GstRtpSsrcDemux *demux;
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gboolean res = FALSE;
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demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_NEWSEGMENT:
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default:
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{
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GSList *walk;
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res = TRUE;
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GST_PAD_LOCK (demux);
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for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
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GstRtpSsrcDemuxPad *pad = (GstRtpSsrcDemuxPad *) walk->data;
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gst_event_ref (event);
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res &= gst_pad_push_event (pad->rtcp_pad, event);
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}
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GST_PAD_UNLOCK (demux);
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gst_event_unref (event);
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break;
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}
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}
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gst_object_unref (demux);
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return res;
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}
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static GstFlowReturn
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gst_rtp_ssrc_demux_chain (GstPad * pad, GstBuffer * buf)
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{
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GstFlowReturn ret;
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GstRtpSsrcDemux *demux;
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guint32 ssrc;
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GstRtpSsrcDemuxPad *dpad;
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demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
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if (!gst_rtp_buffer_validate (buf))
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goto invalid_payload;
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ssrc = gst_rtp_buffer_get_ssrc (buf);
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GST_DEBUG_OBJECT (demux, "received buffer of SSRC %08x", ssrc);
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GST_PAD_LOCK (demux);
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dpad = find_demux_pad_for_ssrc (demux, ssrc);
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if (dpad == NULL) {
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if (!(dpad =
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create_demux_pad_for_ssrc (demux, ssrc,
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GST_BUFFER_TIMESTAMP (buf))))
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goto create_failed;
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}
|
|
GST_PAD_UNLOCK (demux);
|
|
|
|
/* push to srcpad */
|
|
ret = gst_pad_push (dpad->rtp_pad, buf);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
invalid_payload:
|
|
{
|
|
/* this is fatal and should be filtered earlier */
|
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Dropping invalid RTP payload"));
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
create_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Could not create new pad"));
|
|
GST_PAD_UNLOCK (demux);
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_ssrc_demux_rtcp_chain (GstPad * pad, GstBuffer * buf)
|
|
{
|
|
GstFlowReturn ret;
|
|
GstRtpSsrcDemux *demux;
|
|
guint32 ssrc;
|
|
GstRtpSsrcDemuxPad *dpad;
|
|
GstRTCPPacket packet;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (GST_OBJECT_PARENT (pad));
|
|
|
|
if (!gst_rtcp_buffer_validate (buf))
|
|
goto invalid_rtcp;
|
|
|
|
if (!gst_rtcp_buffer_get_first_packet (buf, &packet))
|
|
goto invalid_rtcp;
|
|
|
|
/* first packet must be SR or RR or else the validate would have failed */
|
|
switch (gst_rtcp_packet_get_type (&packet)) {
|
|
case GST_RTCP_TYPE_SR:
|
|
/* get the ssrc so that we can route it to the right source pad */
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL,
|
|
NULL);
|
|
break;
|
|
case GST_RTCP_TYPE_RR:
|
|
ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
|
|
break;
|
|
default:
|
|
goto invalid_rtcp;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (demux, "received RTCP of SSRC %08x", ssrc);
|
|
|
|
GST_PAD_LOCK (demux);
|
|
dpad = find_demux_pad_for_ssrc (demux, ssrc);
|
|
if (dpad == NULL) {
|
|
GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
|
|
if (!(dpad = create_demux_pad_for_ssrc (demux, ssrc, -1)))
|
|
goto create_failed;
|
|
}
|
|
GST_PAD_UNLOCK (demux);
|
|
|
|
/* push to srcpad */
|
|
ret = gst_pad_push (dpad->rtcp_pad, buf);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
invalid_rtcp:
|
|
{
|
|
/* this is fatal and should be filtered earlier */
|
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Dropping invalid RTCP packet"));
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
create_failed:
|
|
{
|
|
GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
|
|
("Could not create new pad"));
|
|
GST_PAD_UNLOCK (demux);
|
|
gst_buffer_unref (buf);
|
|
return GST_FLOW_ERROR;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event)
|
|
{
|
|
GstRtpSsrcDemux *demux;
|
|
gboolean res = FALSE;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
default:
|
|
res = gst_pad_event_default (pad, event);
|
|
break;
|
|
}
|
|
gst_object_unref (demux);
|
|
return res;
|
|
}
|
|
|
|
static GList *
|
|
gst_rtp_ssrc_demux_internal_links (GstPad * pad)
|
|
{
|
|
GstRtpSsrcDemux *demux;
|
|
GList *res = NULL;
|
|
GSList *walk;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
|
|
|
|
GST_PAD_LOCK (demux);
|
|
for (walk = demux->srcpads; walk; walk = g_slist_next (walk)) {
|
|
GstRtpSsrcDemuxPad *dpad = (GstRtpSsrcDemuxPad *) walk->data;
|
|
|
|
if (pad == demux->rtp_sink) {
|
|
res = g_list_prepend (res, dpad->rtp_pad);
|
|
} else if (pad == demux->rtcp_sink) {
|
|
res = g_list_prepend (res, dpad->rtcp_pad);
|
|
} else if (pad == dpad->rtp_pad) {
|
|
res = g_list_prepend (res, demux->rtp_sink);
|
|
break;
|
|
} else if (pad == dpad->rtcp_pad) {
|
|
res = g_list_prepend (res, demux->rtcp_sink);
|
|
break;
|
|
}
|
|
}
|
|
GST_PAD_UNLOCK (demux);
|
|
|
|
gst_object_unref (demux);
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_ssrc_demux_src_query (GstPad * pad, GstQuery * query)
|
|
{
|
|
GstRtpSsrcDemux *demux;
|
|
gboolean res = FALSE;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (gst_pad_get_parent (pad));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
{
|
|
|
|
if ((res = gst_pad_peer_query (demux->rtp_sink, query))) {
|
|
gboolean live;
|
|
GstClockTime min_latency, max_latency;
|
|
GstRtpSsrcDemuxPad *demuxpad;
|
|
|
|
demuxpad = gst_pad_get_element_private (pad);
|
|
|
|
gst_query_parse_latency (query, &live, &min_latency, &max_latency);
|
|
|
|
GST_DEBUG_OBJECT (demux, "peer min latency %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (min_latency));
|
|
|
|
GST_DEBUG_OBJECT (demux,
|
|
"latency for SSRC %08x, latency %" GST_TIME_FORMAT, demuxpad->ssrc,
|
|
GST_TIME_ARGS (demuxpad->first_ts));
|
|
|
|
gst_query_set_latency (query, live, min_latency, max_latency);
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, query);
|
|
break;
|
|
}
|
|
gst_object_unref (demux);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_ssrc_demux_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn ret;
|
|
GstRtpSsrcDemux *demux;
|
|
|
|
demux = GST_RTP_SSRC_DEMUX (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = 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:
|
|
gst_rtp_ssrc_demux_reset (demux);
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
default:
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|