gstreamer/gst/rtsp-server/rtsp-stream-transport.c
Wim Taymans e4ea72ccdf stream: use the address managed by the stream
Use the address managed by the stream for multicast. This allows us to have 1
multicast address for each stream.
Because the address is now managed by the stream we don't have to pass it around
anymore.
Set the address pool on the streams.
2012-11-15 16:18:29 +01:00

231 lines
6.1 KiB
C

/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <string.h>
#include <stdlib.h>
#include <gst/app/gstappsrc.h>
#include <gst/app/gstappsink.h>
#include "rtsp-stream-transport.h"
enum
{
PROP_0,
PROP_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
#define GST_CAT_DEFAULT rtsp_stream_transport_debug
static void gst_rtsp_stream_transport_finalize (GObject * obj);
G_DEFINE_TYPE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
G_TYPE_OBJECT);
static void
gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtsp_stream_transport_finalize;
GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
0, "GstRTSPStreamTransport");
}
static void
gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
{
}
static void
gst_rtsp_stream_transport_finalize (GObject * obj)
{
GstRTSPStreamTransport *trans;
trans = GST_RTSP_STREAM_TRANSPORT (obj);
/* remove callbacks now */
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
if (trans->transport)
gst_rtsp_transport_free (trans->transport);
#if 0
if (trans->rtpsource)
g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL);
#endif
G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
}
/**
* gst_rtsp_stream_transport_new:
* @stream: a #GstRTSPStream
* @tr: (transfer full): a GstRTSPTransport
*
* Create a new #GstRTSPStreamTransport that can be used to manage
* @stream with transport @tr.
*
* Returns: a new #GstRTSPStreamTransport
*/
GstRTSPStreamTransport *
gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
{
GstRTSPStreamTransport *trans;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
g_return_val_if_fail (tr != NULL, NULL);
trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
trans->stream = stream;
trans->transport = tr;
return trans;
}
/**
* gst_rtsp_stream_transport_set_callbacks:
* @trans: a #GstRTSPStreamTransport
* @send_rtp: (scope notified): a callback called when RTP should be sent
* @send_rtcp: (scope notified): a callback called when RTCP should be sent
* @user_data: user data passed to callbacks
* @notify: called with the user_data when no longer needed.
*
* Install callbacks that will be called when data for a stream should be sent
* to a client. This is usually used when sending RTP/RTCP over TCP.
*/
void
gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
gpointer user_data, GDestroyNotify notify)
{
trans->send_rtp = send_rtp;
trans->send_rtcp = send_rtcp;
if (trans->notify)
trans->notify (trans->user_data);
trans->user_data = user_data;
trans->notify = notify;
}
/**
* gst_rtsp_stream_transport_set_keepalive:
* @trans: a #GstRTSPStreamTransport
* @keep_alive: a callback called when the receiver is active
* @user_data: user data passed to callback
* @notify: called with the user_data when no longer needed.
*
* Install callbacks that will be called when RTCP packets are received from the
* receiver of @trans.
*/
void
gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
{
trans->keep_alive = keep_alive;
if (trans->ka_notify)
trans->ka_notify (trans->ka_user_data);
trans->ka_user_data = user_data;
trans->ka_notify = notify;
}
/**
* gst_rtsp_stream_transport_set_transport:
* @trans: a #GstRTSPStreamTransport
* @tr: (transfer full): a client #GstRTSPTransport
*
* Set @tr as the client transport. This function takes ownership of the
* passed @tr.
*/
void
gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
GstRTSPTransport * tr)
{
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
g_return_if_fail (tr != NULL);
/* keep track of the transports in the stream. */
if (trans->transport)
gst_rtsp_transport_free (trans->transport);
trans->transport = tr;
}
/**
* gst_rtsp_stream_transport_send_rtp:
* @trans: a #GstRTSPStreamTransport
* @buffer: a #GstBuffer
*
* Send @buffer to the installed RTP callback for @trans.
*
* Returns: %TRUE on success
*/
gboolean
gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
gboolean res = FALSE;
if (trans->send_rtp)
res =
trans->send_rtp (buffer, trans->transport->interleaved.min,
trans->user_data);
return res;
}
/**
* gst_rtsp_stream_transport_send_rtcp:
* @trans: a #GstRTSPStreamTransport
* @buffer: a #GstBuffer
*
* Send @buffer to the installed RTCP callback for @trans.
*
* Returns: %TRUE on success
*/
gboolean
gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
gboolean res = FALSE;
if (trans->send_rtcp)
res =
trans->send_rtcp (buffer, trans->transport->interleaved.max,
trans->user_data);
return res;
}
/**
* gst_rtsp_stream_transport_keep_alive:
* @trans: a #GstRTSPStreamTransport
*
* Signal the installed keep_alive callback for @trans.
*/
void
gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
{
if (trans->keep_alive)
trans->keep_alive (trans->ka_user_data);
}