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45b6693b39
Return a boxed GstRTSPAddress from the GstRTSPAddressPool. This allows us to store more info in the structure and allows us to more easily return the address to the right pool when no longer needed. Pass the address to the StreamTransport so that we can return it to the pool when the stream transport is freed or changed.
240 lines
6.5 KiB
C
240 lines
6.5 KiB
C
/* 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, 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 (trans->addr)
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gst_rtsp_address_free (trans->addr);
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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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* @tr: (transfer full): a GstRTSPTransport
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* @addr: (transfer full) (allow none): an optional GstRTSPAddress
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*
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* Create a new #GstRTSPStreamTransport that can be used to manage
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* @stream with transport @tr.
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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, GstRTSPTransport * tr,
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GstRTSPAddress * addr)
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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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g_return_val_if_fail (tr != NULL, 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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trans->transport = tr;
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trans->addr = addr;
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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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* @tr: (transfer full): a client #GstRTSPTransport
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* @addr: (transfer full) (allow none): a ##GstRTSPAddress
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*
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* Set @tr and the optional @addr as the client transport. This function
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* takes ownership of the passed @tr and @addr.
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*/
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void
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gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
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GstRTSPTransport * tr, GstRTSPAddress * addr)
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{
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g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
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g_return_if_fail (tr != 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 = tr;
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if (trans->addr)
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gst_rtsp_address_free (trans->addr);
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trans->addr = addr;
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}
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/**
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* gst_rtsp_stream_transport_send_rtp:
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* @trans: a #GstRTSPStreamTransport
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* @buffer: a #GstBuffer
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*
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* Send @buffer to the installed RTP callback for @trans.
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*
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* Returns: %TRUE on success
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*/
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gboolean
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gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
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GstBuffer * buffer)
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{
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gboolean res = FALSE;
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if (trans->send_rtp)
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res =
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trans->send_rtp (buffer, trans->transport->interleaved.min,
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trans->user_data);
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return res;
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}
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/**
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* gst_rtsp_stream_transport_send_rtcp:
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* @trans: a #GstRTSPStreamTransport
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* @buffer: a #GstBuffer
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*
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* Send @buffer to the installed RTCP callback for @trans.
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*
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* Returns: %TRUE on success
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*/
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gboolean
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gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
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GstBuffer * buffer)
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{
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gboolean res = FALSE;
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if (trans->send_rtcp)
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res =
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trans->send_rtcp (buffer, trans->transport->interleaved.max,
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trans->user_data);
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return res;
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}
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/**
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* gst_rtsp_stream_transport_keep_alive:
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* @trans: a #GstRTSPStreamTransport
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*
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* Signal the installed keep_alive callback for @trans.
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*/
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void
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gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
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{
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if (trans->keep_alive)
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trans->keep_alive (trans->ka_user_data);
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}
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