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203 lines
5.7 KiB
C
203 lines
5.7 KiB
C
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/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at 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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#include <string.h>
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#include <stdlib.h>
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#include <gst/app/gstappsrc.h>
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#include <gst/app/gstappsink.h>
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#include "rtsp-stream-transport.h"
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enum
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{
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PROP_0,
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PROP_LAST
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};
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GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
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#define GST_CAT_DEFAULT rtsp_stream_transport_debug
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static void gst_rtsp_stream_transport_finalize (GObject * obj);
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G_DEFINE_TYPE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
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G_TYPE_OBJECT);
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static void
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gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = G_OBJECT_CLASS (klass);
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gobject_class->finalize = gst_rtsp_stream_transport_finalize;
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GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
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0, "GstRTSPStreamTransport");
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}
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static void
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gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
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{
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}
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static void
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gst_rtsp_stream_transport_finalize (GObject * obj)
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{
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GstRTSPStreamTransport *trans;
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trans = GST_RTSP_STREAM_TRANSPORT (obj);
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/* remove callbacks now */
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gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
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gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
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if (trans->transport)
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gst_rtsp_transport_free (trans->transport);
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#if 0
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if (trans->rtpsource)
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g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL);
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#endif
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G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
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}
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/**
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* gst_rtsp_stream_transport_new:
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* @stream: a #GstRTSPStream
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*
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* Create a new #GstRTSPStreamTransport that can be used for
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* @stream.
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*
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* Returns: a new #GstRTSPStreamTransport
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*/
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GstRTSPStreamTransport *
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gst_rtsp_stream_transport_new (GstRTSPStream * stream)
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{
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GstRTSPStreamTransport *trans;
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g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
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trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
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trans->stream = stream;
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return trans;
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}
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/**
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* gst_rtsp_stream_transport_set_callbacks:
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* @trans: a #GstRTSPStreamTransport
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* @send_rtp: (scope notified): a callback called when RTP should be sent
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* @send_rtcp: (scope notified): a callback called when RTCP should be sent
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* @user_data: user data passed to callbacks
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* @notify: called with the user_data when no longer needed.
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*
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* Install callbacks that will be called when data for a stream should be sent
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* to a client. This is usually used when sending RTP/RTCP over TCP.
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*/
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void
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gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
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GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
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gpointer user_data, GDestroyNotify notify)
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{
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trans->send_rtp = send_rtp;
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trans->send_rtcp = send_rtcp;
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if (trans->notify)
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trans->notify (trans->user_data);
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trans->user_data = user_data;
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trans->notify = notify;
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}
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/**
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* gst_rtsp_stream_transport_set_keepalive:
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* @trans: a #GstRTSPStreamTransport
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* @keep_alive: a callback called when the receiver is active
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* @user_data: user data passed to callback
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* @notify: called with the user_data when no longer needed.
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*
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* Install callbacks that will be called when RTCP packets are received from the
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* receiver of @trans.
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*/
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void
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gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
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GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
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{
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trans->keep_alive = keep_alive;
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if (trans->ka_notify)
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trans->ka_notify (trans->ka_user_data);
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trans->ka_user_data = user_data;
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trans->ka_notify = notify;
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}
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/**
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* gst_rtsp_stream_transport_set_transport:
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* @trans: a #GstRTSPStreamTransport
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* @ct: a client #GstRTSPTransport
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*
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* Set @ct as the client transport and create and return a matching server
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* transport. This function takes ownership of the passed @ct.
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*
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* Returns: a server transport or NULL if something went wrong. Use
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* gst_rtsp_transport_free () after usage.
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*/
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GstRTSPTransport *
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gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
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GstRTSPTransport * ct)
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{
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GstRTSPTransport *st;
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g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
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g_return_val_if_fail (ct != NULL, NULL);
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/* prepare the server transport */
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gst_rtsp_transport_new (&st);
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st->trans = ct->trans;
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st->profile = ct->profile;
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st->lower_transport = ct->lower_transport;
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switch (st->lower_transport) {
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case GST_RTSP_LOWER_TRANS_UDP:
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st->client_port = ct->client_port;
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st->server_port = trans->stream->server_port;
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break;
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case GST_RTSP_LOWER_TRANS_UDP_MCAST:
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ct->port = st->port = trans->stream->server_port;
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st->destination = g_strdup (ct->destination);
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st->ttl = ct->ttl;
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break;
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case GST_RTSP_LOWER_TRANS_TCP:
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st->interleaved = ct->interleaved;
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default:
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break;
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}
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if (trans->stream->session)
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g_object_get (trans->stream->session, "internal-ssrc", &st->ssrc, NULL);
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/* keep track of the transports in the stream. */
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if (trans->transport)
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gst_rtsp_transport_free (trans->transport);
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trans->transport = ct;
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return st;
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}
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