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Original commit message from CVS: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream), (gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp), (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpclient.c: (create_stream), (gst_rtp_client_request_new_pad): * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop): * gst/rtpmanager/gstrtpmanager.c: (plugin_init): * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init), (gst_rtp_session_request_new_pad): * gst/rtpmanager/gstrtpssrcdemux.c: Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
354 lines
9.3 KiB
C
354 lines
9.3 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim@fluendo.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 "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_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 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@fluendo.com>"
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};
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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_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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/* srcpad stuff */
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static gboolean gst_rtp_ssrc_demux_src_event (GstPad * pad, GstEvent * event);
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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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GstPad *pad;
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guint32 ssrc;
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GstCaps *caps;
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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 GstPad *
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find_rtp_pad_for_ssrc (GstRTPSsrcDemux * demux, guint32 ssrc)
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{
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GSList *walk;
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for (walk = demux->rtp_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->pad;
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}
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return NULL;
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}
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static GstPad *
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create_rtp_pad_for_ssrc (GstRTPSsrcDemux * demux, guint32 ssrc)
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{
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GstPad *result;
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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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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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result = gst_pad_new_from_template (templ, padname);
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g_free (padname);
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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->pad = result;
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demux->rtp_srcpads = g_slist_prepend (demux->rtp_srcpads, demuxpad);
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/* copy caps from input */
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gst_pad_set_caps (result, GST_PAD_CAPS (demux->rtp_sink));
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gst_pad_set_event_function (result, gst_rtp_ssrc_demux_src_event);
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gst_pad_set_active (result, TRUE);
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gst_element_add_pad (GST_ELEMENT_CAST (demux), result);
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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, result);
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return result;
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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_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->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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}
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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_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_NEWSEGMENT:
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default:
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res = gst_pad_event_default (pad, event);
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break;
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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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GstPad *srcpad;
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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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srcpad = find_rtp_pad_for_ssrc (demux, ssrc);
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if (srcpad == NULL) {
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GST_DEBUG_OBJECT (demux, "creating pad for SSRC %08x", ssrc);
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srcpad = create_rtp_pad_for_ssrc (demux, ssrc);
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if (!srcpad)
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goto create_failed;
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}
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/* push to srcpad */
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ret = gst_pad_push (srcpad, buf);
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return ret;
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/* ERRORS */
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invalid_payload:
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{
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/* this is fatal and should be filtered earlier */
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GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
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("Dropping invalid RTP payload"));
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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}
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create_failed:
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{
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/* this is not fatal yet */
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GST_ELEMENT_ERROR (demux, STREAM, DECODE, (NULL),
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("Could not create new pad"));
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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}
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}
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static gboolean
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gst_rtp_ssrc_demux_src_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_SEEK:
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default:
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res = gst_pad_event_default (pad, event);
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break;
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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 GstStateChangeReturn
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gst_rtp_ssrc_demux_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstRTPSsrcDemux *demux;
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demux = GST_RTP_SSRC_DEMUX (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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case GST_STATE_CHANGE_READY_TO_NULL:
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default:
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break;
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}
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return ret;
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}
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