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.
21 #include "rtsp-session.h"
25 #define DEFAULT_TIMEOUT 60
33 GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug);
34 #define GST_CAT_DEFAULT rtsp_session_media_debug
36 static void gst_rtsp_session_media_finalize (GObject * obj);
38 G_DEFINE_TYPE (GstRTSPSessionMedia, gst_rtsp_session_media, G_TYPE_OBJECT);
41 gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass)
43 GObjectClass *gobject_class;
45 gobject_class = G_OBJECT_CLASS (klass);
47 gobject_class->finalize = gst_rtsp_session_media_finalize;
49 GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0,
50 "GstRTSPSessionMedia");
54 gst_rtsp_session_media_init (GstRTSPSessionMedia * media)
56 g_mutex_init (&media->lock);
57 media->state = GST_RTSP_STATE_INIT;
61 gst_rtsp_session_media_finalize (GObject * obj)
63 GstRTSPSessionMedia *media;
65 media = GST_RTSP_SESSION_MEDIA (obj);
67 GST_INFO ("free session media %p", media);
69 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
71 g_ptr_array_unref (media->transports);
73 gst_rtsp_url_free (media->url);
74 g_object_unref (media->media);
75 g_mutex_clear (&media->lock);
77 G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj);
81 free_session_media (gpointer data)
84 g_object_unref (data);
88 * gst_rtsp_session_media_new:
89 * @url: the #GstRTSPUrl
90 * @media: the #GstRTSPMedia
92 * Create a new #GstRTPSessionMedia that manages the streams
93 * in @media for @url. @media should be prepared.
95 * Ownership is taken of @media.
97 * Returns: a new #GstRTSPSessionMedia.
100 gst_rtsp_session_media_new (const GstRTSPUrl * url, GstRTSPMedia * media)
102 GstRTSPSessionMedia *result;
105 g_return_val_if_fail (url != NULL, NULL);
106 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
107 g_return_val_if_fail (media->status == GST_RTSP_MEDIA_STATUS_PREPARED, NULL);
109 result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
110 result->url = gst_rtsp_url_copy ((GstRTSPUrl *) url);
111 result->media = media;
113 /* prealloc the streams now, filled with NULL */
114 n_streams = gst_rtsp_media_n_streams (media);
115 result->transports = g_ptr_array_new_full (n_streams, free_session_media);
116 g_ptr_array_set_size (result->transports, n_streams);
122 * gst_rtsp_session_media_set_transport:
123 * @media: a #GstRTSPSessionMedia
124 * @stream: a #GstRTSPStream
125 * @tr: a #GstRTSPTransport
126 * @addr: (transfer full) (allow none): an optional #GstRTSPAddress
128 * Configure the transport for @stream to @tr in @media.
130 * Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream.
132 GstRTSPStreamTransport *
133 gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
134 GstRTSPStream * stream, GstRTSPTransport * tr, GstRTSPAddress * addr)
136 GstRTSPStreamTransport *result;
138 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
139 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
140 g_return_val_if_fail (stream->idx < media->transports->len, NULL);
142 g_mutex_lock (&media->lock);
143 result = g_ptr_array_index (media->transports, stream->idx);
144 if (result == NULL) {
145 result = gst_rtsp_stream_transport_new (stream, tr, addr);
146 g_ptr_array_index (media->transports, stream->idx) = result;
147 g_mutex_unlock (&media->lock);
149 gst_rtsp_stream_transport_set_transport (result, tr, addr);
150 g_mutex_unlock (&media->lock);
157 * gst_rtsp_session_media_get_transport:
158 * @media: a #GstRTSPSessionMedia
159 * @idx: the stream index
161 * Get a previously created #GstRTSPStreamTransport for the stream at @idx.
163 * Returns: (transfer none): a #GstRTSPStreamTransport that is valid until the
164 * session of @media is unreffed.
166 GstRTSPStreamTransport *
167 gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
169 GstRTSPStreamTransport *result;
171 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
172 g_return_val_if_fail (idx < media->transports->len, NULL);
174 g_mutex_lock (&media->lock);
175 result = g_ptr_array_index (media->transports, idx);
176 g_mutex_unlock (&media->lock);
182 * gst_rtsp_session_media_alloc_channels:
183 * @media: a #GstRTSPSessionMedia
184 * @range: a #GstRTSPRange
186 * Fill @range with the next available min and max channels for
187 * interleaved transport.
189 * Returns: %TRUE on success.
192 gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
193 GstRTSPRange * range)
195 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
197 g_mutex_lock (&media->lock);
198 range->min = media->counter++;
199 range->max = media->counter++;
200 g_mutex_unlock (&media->lock);
206 * gst_rtsp_session_media_set_state:
207 * @media: a #GstRTSPSessionMedia
208 * @state: the new state
210 * Tell the media object @media to change to @state.
212 * Returns: %TRUE on success.
215 gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state)
219 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
221 g_mutex_lock (&media->lock);
222 ret = gst_rtsp_media_set_state (media->media, state, media->transports);
223 g_mutex_unlock (&media->lock);