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Original commit message from CVS: * gst/deinterlace/gstdeinterlace.c: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpclient.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/sdp/gstsdpdemux.c: More doc updates. More xrefs.
484 lines
13 KiB
C
484 lines
13 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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-gstrtpclient
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* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpsession
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*
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* This element handles RTP data from one client. It accepts multiple RTP streams that
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* should be synchronized together.
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*
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* Normally the SSRCs that map to the same CNAME (as given in the RTCP SDES messages)
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* should be synchronized.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* FIXME: gst-launch
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* ]| FIXME: describe
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* </refsect2>
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*
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* Last reviewed on 2007-04-02 (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 <stdlib.h>
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#include <string.h>
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#include "gstrtpclient.h"
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/* elementfactory information */
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static const GstElementDetails rtpclient_details =
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GST_ELEMENT_DETAILS ("RTP Client",
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"Filter/Network/RTP",
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"Implement an RTP client",
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"Wim Taymans <wim.taymans@gmail.com>");
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/* sink pads */
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static GstStaticPadTemplate rtpclient_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("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 rtpclient_sync_sink_template =
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GST_STATIC_PAD_TEMPLATE ("sync_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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/* src pads */
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static GstStaticPadTemplate rtpclient_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("rtp_src_%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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#define GST_RTP_CLIENT_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_CLIENT, GstRtpClientPrivate))
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struct _GstRtpClientPrivate
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{
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gint foo;
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};
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/* all the info needed to handle the stream with SSRC */
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typedef struct
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{
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GstRtpClient *client;
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/* the SSRC of this stream */
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guint32 ssrc;
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/* RTP and RTCP in */
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GstPad *rtp_sink;
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GstPad *sync_sink;
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/* the jitterbuffer */
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GstElement *jitterbuffer;
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/* the payload demuxer */
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GstElement *ptdemux;
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/* the new-pad signal */
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gulong new_pad_sig;
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} GstRtpClientStream;
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/* the PT demuxer found a new payload type */
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static void
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new_pad (GstElement * element, GstPad * pad, GstRtpClientStream * stream)
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{
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}
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/* create a new stream for SSRC.
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*
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* We create a jitterbuffer and an payload demuxer for the SSRC. The sinkpad of
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* the jitterbuffer is ghosted to the bin. We connect a pad-added signal to
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* rtpptdemux so that we can ghost the payload pads outside.
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*
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* +-----------------+ +---------------+
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* | rtpjitterbuffer | | rtpptdemux |
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* +- sink src - sink |
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* / +-----------------+ +---------------+
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*
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*/
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static GstRtpClientStream *
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create_stream (GstRtpClient * rtpclient, guint32 ssrc)
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{
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GstRtpClientStream *stream;
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gchar *name;
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GstPad *srcpad, *sinkpad;
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GstPadLinkReturn res;
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stream = g_new0 (GstRtpClientStream, 1);
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stream->ssrc = ssrc;
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stream->client = rtpclient;
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stream->jitterbuffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL);
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if (!stream->jitterbuffer)
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goto no_jitterbuffer;
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stream->ptdemux = gst_element_factory_make ("gstrtpptdemux", NULL);
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if (!stream->ptdemux)
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goto no_ptdemux;
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/* add elements to bin */
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gst_bin_add (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
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gst_bin_add (GST_BIN_CAST (rtpclient), stream->ptdemux);
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/* link jitterbuffer and PT demuxer */
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srcpad = gst_element_get_static_pad (stream->jitterbuffer, "src");
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sinkpad = gst_element_get_static_pad (stream->ptdemux, "sink");
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res = gst_pad_link (srcpad, sinkpad);
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gst_object_unref (srcpad);
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gst_object_unref (sinkpad);
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if (res != GST_PAD_LINK_OK)
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goto could_not_link;
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/* add stream to list */
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rtpclient->streams = g_list_prepend (rtpclient->streams, stream);
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/* ghost sinkpad */
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name = g_strdup_printf ("rtp_sink_%d", ssrc);
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sinkpad = gst_element_get_static_pad (stream->jitterbuffer, "sink");
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stream->rtp_sink = gst_ghost_pad_new (name, sinkpad);
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gst_object_unref (sinkpad);
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g_free (name);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->rtp_sink);
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/* add signal to ptdemuxer */
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stream->new_pad_sig =
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g_signal_connect (G_OBJECT (stream->ptdemux), "pad-added",
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G_CALLBACK (new_pad), stream);
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return stream;
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/* ERRORS */
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no_jitterbuffer:
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{
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g_free (stream);
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g_warning ("gstrtpclient: could not create gstrtpjitterbuffer element");
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return NULL;
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}
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no_ptdemux:
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{
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gst_object_unref (stream->jitterbuffer);
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g_free (stream);
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g_warning ("gstrtpclient: could not create gstrtpptdemux element");
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return NULL;
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}
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could_not_link:
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{
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gst_bin_remove (GST_BIN_CAST (rtpclient), stream->jitterbuffer);
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gst_bin_remove (GST_BIN_CAST (rtpclient), stream->ptdemux);
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g_free (stream);
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g_warning ("gstrtpclient: could not link jitterbuffer and ptdemux 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 void
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free_stream (GstRtpClientStream * stream)
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{
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gst_object_unref (stream->jitterbuffer);
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g_free (stream);
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}
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#endif
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/* find the stream for the given SSRC, return NULL if the stream did not exist
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*/
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static GstRtpClientStream *
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find_stream_by_ssrc (GstRtpClient * client, guint32 ssrc)
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{
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GstRtpClientStream *stream;
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GList *walk;
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for (walk = client->streams; walk; walk = g_list_next (walk)) {
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stream = (GstRtpClientStream *) 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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/* signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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/* GObject vmethods */
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static void gst_rtp_client_finalize (GObject * object);
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static void gst_rtp_client_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_client_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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/* GstElement vmethods */
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static GstStateChangeReturn gst_rtp_client_change_state (GstElement * element,
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GstStateChange transition);
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static GstPad *gst_rtp_client_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name);
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static void gst_rtp_client_release_pad (GstElement * element, GstPad * pad);
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/*static guint gst_rtp_client_signals[LAST_SIGNAL] = { 0 }; */
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GST_BOILERPLATE (GstRtpClient, gst_rtp_client, GstBin, GST_TYPE_BIN);
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static void
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gst_rtp_client_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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/* sink pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpclient_rtp_sink_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpclient_sync_sink_template));
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/* src pads */
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&rtpclient_rtp_src_template));
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gst_element_class_set_details (element_class, &rtpclient_details);
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}
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static void
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gst_rtp_client_class_init (GstRtpClientClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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g_type_class_add_private (klass, sizeof (GstRtpClientPrivate));
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gobject_class->finalize = gst_rtp_client_finalize;
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gobject_class->set_property = gst_rtp_client_set_property;
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gobject_class->get_property = gst_rtp_client_get_property;
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_client_change_state);
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gstelement_class->request_new_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_client_request_new_pad);
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gstelement_class->release_pad =
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GST_DEBUG_FUNCPTR (gst_rtp_client_release_pad);
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}
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static void
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gst_rtp_client_init (GstRtpClient * rtpclient, GstRtpClientClass * klass)
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{
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rtpclient->priv = GST_RTP_CLIENT_GET_PRIVATE (rtpclient);
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}
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static void
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gst_rtp_client_finalize (GObject * object)
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{
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GstRtpClient *rtpclient;
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rtpclient = GST_RTP_CLIENT (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static void
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gst_rtp_client_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpClient *rtpclient;
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rtpclient = GST_RTP_CLIENT (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_client_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpClient *rtpclient;
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rtpclient = GST_RTP_CLIENT (object);
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switch (prop_id) {
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static GstStateChangeReturn
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gst_rtp_client_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn res;
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GstRtpClient *rtpclient;
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rtpclient = GST_RTP_CLIENT (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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break;
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default:
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break;
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}
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res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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switch (transition) {
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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break;
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case GST_STATE_CHANGE_READY_TO_NULL:
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break;
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default:
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break;
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}
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return res;
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}
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/* We have 2 request pads (rtp_sink_%d and sync_sink_%d), the %d is assumed to
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* be the SSRC of the stream.
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*
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* We require that the rtp pad is requested first for a particular SSRC, then
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* (optionaly) the sync pad can be requested. If no sync pad is requested, no
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* sync information can be exchanged for this stream.
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*/
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static GstPad *
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gst_rtp_client_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name)
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{
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GstRtpClient *rtpclient;
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GstElementClass *klass;
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GstPadTemplate *rtp_sink_templ, *sync_sink_templ;
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guint32 ssrc;
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GstRtpClientStream *stream;
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GstPad *result;
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g_return_val_if_fail (templ != NULL, NULL);
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g_return_val_if_fail (GST_IS_RTP_CLIENT (element), NULL);
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if (templ->direction != GST_PAD_SINK)
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goto wrong_direction;
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rtpclient = GST_RTP_CLIENT (element);
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klass = GST_ELEMENT_GET_CLASS (element);
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/* figure out the template */
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rtp_sink_templ = gst_element_class_get_pad_template (klass, "rtp_sink_%d");
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sync_sink_templ = gst_element_class_get_pad_template (klass, "sync_sink_%d");
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if (templ != rtp_sink_templ && templ != sync_sink_templ)
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goto wrong_template;
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if (templ == rtp_sink_templ) {
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/* create new rtp sink pad. If a stream with the pad number already exists
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* we have an error, else we create the sinkpad, add a jitterbuffer and
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* ptdemuxer. */
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if (name == NULL || strlen (name) < 9)
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goto no_name;
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ssrc = atoi (&name[9]);
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/* see if a stream with that name exists, if so we have an error. */
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stream = find_stream_by_ssrc (rtpclient, ssrc);
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if (stream != NULL)
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goto stream_exists;
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/* ok, create new stream */
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stream = create_stream (rtpclient, ssrc);
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if (stream == NULL)
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goto stream_not_found;
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result = stream->rtp_sink;
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} else {
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/* create new rtp sink pad. We can only do this if the RTP pad was
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* requested before, meaning the session with the padnumber must exist. */
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if (name == NULL || strlen (name) < 10)
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goto no_name;
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ssrc = atoi (&name[10]);
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/* find stream */
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stream = find_stream_by_ssrc (rtpclient, ssrc);
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if (stream == NULL)
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goto stream_not_found;
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stream->sync_sink =
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gst_pad_new_from_static_template (&rtpclient_sync_sink_template, name);
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gst_element_add_pad (GST_ELEMENT_CAST (rtpclient), stream->sync_sink);
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result = stream->sync_sink;
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}
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return result;
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/* ERRORS */
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wrong_direction:
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{
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g_warning ("gstrtpclient: request pad that is not a SINK pad");
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return NULL;
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}
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wrong_template:
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{
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g_warning ("gstrtpclient: this is not our template");
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return NULL;
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}
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no_name:
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{
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g_warning ("gstrtpclient: no padname was specified");
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return NULL;
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}
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stream_exists:
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{
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g_warning ("gstrtpclient: stream with SSRC %d already registered", ssrc);
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return NULL;
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}
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stream_not_found:
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{
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g_warning ("gstrtpclient: stream with SSRC %d not yet registered", ssrc);
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return NULL;
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}
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}
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static void
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gst_rtp_client_release_pad (GstElement * element, GstPad * pad)
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{
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}
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