2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 * SECTION:rtsp-stream-transport
21 * @short_description: A media stream transport configuration
22 * @see_also: #GstRTSPStream, #GstRTSPSessionMedia
24 * The #GstRTSPStreamTransport configures the transport used by a
25 * #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
27 * With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
28 * to handle the RTP and RTCP packets from the stream, for example when they
29 * need to be sent over TCP.
31 * With gst_rtsp_stream_transport_set_active() the transports are added and
32 * removed from the stream.
34 * A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
35 * is received from the client. It will also call
36 * gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
38 * Last reviewed on 2013-07-16 (1.0.0)
44 #include "rtsp-stream-transport.h"
46 #define GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE(obj) \
47 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportPrivate))
49 struct _GstRTSPStreamTransportPrivate
51 GstRTSPStream *stream;
53 GstRTSPSendFunc send_rtp;
54 GstRTSPSendFunc send_rtcp;
56 GDestroyNotify notify;
58 GstRTSPKeepAliveFunc keep_alive;
59 gpointer ka_user_data;
60 GDestroyNotify ka_notify;
64 GstRTSPTransport *transport;
76 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
77 #define GST_CAT_DEFAULT rtsp_stream_transport_debug
79 static void gst_rtsp_stream_transport_finalize (GObject * obj);
81 G_DEFINE_TYPE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
85 gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
87 GObjectClass *gobject_class;
89 g_type_class_add_private (klass, sizeof (GstRTSPStreamTransportPrivate));
91 gobject_class = G_OBJECT_CLASS (klass);
93 gobject_class->finalize = gst_rtsp_stream_transport_finalize;
95 GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
96 0, "GstRTSPStreamTransport");
100 gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
102 GstRTSPStreamTransportPrivate *priv =
103 GST_RTSP_STREAM_TRANSPORT_GET_PRIVATE (trans);
109 gst_rtsp_stream_transport_finalize (GObject * obj)
111 GstRTSPStreamTransportPrivate *priv;
112 GstRTSPStreamTransport *trans;
114 trans = GST_RTSP_STREAM_TRANSPORT (obj);
117 /* remove callbacks now */
118 gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
119 gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
122 gst_rtsp_transport_free (priv->transport);
124 G_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
128 * gst_rtsp_stream_transport_new:
129 * @stream: a #GstRTSPStream
130 * @tr: (transfer full): a GstRTSPTransport
132 * Create a new #GstRTSPStreamTransport that can be used to manage
133 * @stream with transport @tr.
135 * Returns: a new #GstRTSPStreamTransport
137 GstRTSPStreamTransport *
138 gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
140 GstRTSPStreamTransportPrivate *priv;
141 GstRTSPStreamTransport *trans;
143 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
144 g_return_val_if_fail (tr != NULL, NULL);
146 trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
148 priv->stream = stream;
149 priv->transport = tr;
155 * gst_rtsp_stream_transport_get_stream:
156 * @trans: a #GstRTSPStreamTransport
158 * Get the #GstRTSPStream used when constructing @trans.
160 * Returns: (transfer none): the stream used when constructing @trans.
163 gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
165 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
167 return trans->priv->stream;
171 * gst_rtsp_stream_transport_set_callbacks:
172 * @trans: a #GstRTSPStreamTransport
173 * @send_rtp: (scope notified): a callback called when RTP should be sent
174 * @send_rtcp: (scope notified): a callback called when RTCP should be sent
175 * @user_data: user data passed to callbacks
176 * @notify: called with the user_data when no longer needed.
178 * Install callbacks that will be called when data for a stream should be sent
179 * to a client. This is usually used when sending RTP/RTCP over TCP.
182 gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
183 GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
184 gpointer user_data, GDestroyNotify notify)
186 GstRTSPStreamTransportPrivate *priv;
188 g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
192 priv->send_rtp = send_rtp;
193 priv->send_rtcp = send_rtcp;
195 priv->notify (priv->user_data);
196 priv->user_data = user_data;
197 priv->notify = notify;
201 * gst_rtsp_stream_transport_set_keepalive:
202 * @trans: a #GstRTSPStreamTransport
203 * @keep_alive: a callback called when the receiver is active
204 * @user_data: user data passed to callback
205 * @notify: called with the user_data when no longer needed.
207 * Install callbacks that will be called when RTCP packets are received from the
208 * receiver of @trans.
211 gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
212 GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
214 GstRTSPStreamTransportPrivate *priv;
216 g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
220 priv->keep_alive = keep_alive;
222 priv->ka_notify (priv->ka_user_data);
223 priv->ka_user_data = user_data;
224 priv->ka_notify = notify;
229 * gst_rtsp_stream_transport_set_transport:
230 * @trans: a #GstRTSPStreamTransport
231 * @tr: (transfer full): a client #GstRTSPTransport
233 * Set @tr as the client transport. This function takes ownership of the
237 gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans,
238 GstRTSPTransport * tr)
240 GstRTSPStreamTransportPrivate *priv;
242 g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
243 g_return_if_fail (tr != NULL);
247 /* keep track of the transports in the stream. */
249 gst_rtsp_transport_free (priv->transport);
250 priv->transport = tr;
254 * gst_rtsp_stream_transport_get_transport:
255 * @trans: a #GstRTSPStreamTransport
257 * Get the transport configured in @trans.
259 * Returns: (transfer none): the transport configured in @trans. It remains
260 * valid for as long as @trans is valid.
262 const GstRTSPTransport *
263 gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
265 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
267 return trans->priv->transport;
271 * gst_rtsp_stream_transport_set_url:
272 * @trans: a #GstRTSPStreamTransport
273 * @url: (transfer none): a client #GstRTSPUrl
275 * Set @url as the client url.
278 gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
279 const GstRTSPUrl * url)
281 GstRTSPStreamTransportPrivate *priv;
283 g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
287 /* keep track of the transports in the stream. */
289 gst_rtsp_url_free (priv->url);
290 priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
294 * gst_rtsp_stream_transport_get_url:
295 * @trans: a #GstRTSPStreamTransport
297 * Get the url configured in @trans.
299 * Returns: (transfer none): the url configured in @trans. It remains
300 * valid for as long as @trans is valid.
303 gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
305 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
307 return trans->priv->url;
311 * gst_rtsp_stream_transport_set_active:
312 * @trans: a #GstRTSPStreamTransport
313 * @active: new state of @trans
315 * Activate or deactivate datatransfer configured in @trans.
317 * Returns: %TRUE when the state was changed.
320 gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
323 GstRTSPStreamTransportPrivate *priv;
326 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
330 if (priv->active == active)
334 res = gst_rtsp_stream_add_transport (priv->stream, trans);
336 res = gst_rtsp_stream_remove_transport (priv->stream, trans);
339 priv->active = active;
345 * gst_rtsp_stream_transport_set_timed_out:
346 * @trans: a #GstRTSPStreamTransport
347 * @timedout: timed out value
349 * Set the timed out state of @trans to @timedout
352 gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
355 g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
357 trans->priv->timed_out = timedout;
361 * gst_rtsp_stream_transport_is_timed_out:
362 * @trans: a #GstRTSPStreamTransport
364 * Check if @trans is timed out.
366 * Returns: %TRUE if @trans timed out.
369 gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
371 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
373 return trans->priv->timed_out;
377 * gst_rtsp_stream_transport_send_rtp:
378 * @trans: a #GstRTSPStreamTransport
379 * @buffer: a #GstBuffer
381 * Send @buffer to the installed RTP callback for @trans.
383 * Returns: %TRUE on success
386 gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
389 GstRTSPStreamTransportPrivate *priv;
390 gboolean res = FALSE;
396 priv->send_rtp (buffer, priv->transport->interleaved.min,
403 * gst_rtsp_stream_transport_send_rtcp:
404 * @trans: a #GstRTSPStreamTransport
405 * @buffer: a #GstBuffer
407 * Send @buffer to the installed RTCP callback for @trans.
409 * Returns: %TRUE on success
412 gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
415 GstRTSPStreamTransportPrivate *priv;
416 gboolean res = FALSE;
422 priv->send_rtcp (buffer, priv->transport->interleaved.max,
429 * gst_rtsp_stream_transport_keep_alive:
430 * @trans: a #GstRTSPStreamTransport
432 * Signal the installed keep_alive callback for @trans.
435 gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
437 GstRTSPStreamTransportPrivate *priv;
441 if (priv->keep_alive)
442 priv->keep_alive (priv->ka_user_data);