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"
29 GST_DEBUG_CATEGORY_STATIC (rtsp_session_media_debug);
30 #define GST_CAT_DEFAULT rtsp_session_media_debug
32 static void gst_rtsp_session_media_finalize (GObject * obj);
34 G_DEFINE_TYPE (GstRTSPSessionMedia, gst_rtsp_session_media, G_TYPE_OBJECT);
37 gst_rtsp_session_media_class_init (GstRTSPSessionMediaClass * klass)
39 GObjectClass *gobject_class;
41 gobject_class = G_OBJECT_CLASS (klass);
43 gobject_class->finalize = gst_rtsp_session_media_finalize;
45 GST_DEBUG_CATEGORY_INIT (rtsp_session_media_debug, "rtspsessionmedia", 0,
46 "GstRTSPSessionMedia");
50 gst_rtsp_session_media_init (GstRTSPSessionMedia * media)
52 g_mutex_init (&media->lock);
53 media->state = GST_RTSP_STATE_INIT;
57 gst_rtsp_session_media_finalize (GObject * obj)
59 GstRTSPSessionMedia *media;
61 media = GST_RTSP_SESSION_MEDIA (obj);
63 GST_INFO ("free session media %p", media);
65 gst_rtsp_session_media_set_state (media, GST_STATE_NULL);
67 g_ptr_array_unref (media->transports);
69 gst_rtsp_url_free (media->url);
70 g_object_unref (media->media);
71 g_mutex_clear (&media->lock);
73 G_OBJECT_CLASS (gst_rtsp_session_media_parent_class)->finalize (obj);
77 free_session_media (gpointer data)
80 g_object_unref (data);
84 * gst_rtsp_session_media_new:
85 * @url: the #GstRTSPUrl
86 * @media: the #GstRTSPMedia
88 * Create a new #GstRTPSessionMedia that manages the streams
89 * in @media for @url. @media should be prepared.
91 * Ownership is taken of @media.
93 * Returns: a new #GstRTSPSessionMedia.
96 gst_rtsp_session_media_new (const GstRTSPUrl * url, GstRTSPMedia * media)
98 GstRTSPSessionMedia *result;
101 g_return_val_if_fail (url != NULL, NULL);
102 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
103 g_return_val_if_fail (media->status == GST_RTSP_MEDIA_STATUS_PREPARED, NULL);
105 result = g_object_new (GST_TYPE_RTSP_SESSION_MEDIA, NULL);
106 result->url = gst_rtsp_url_copy ((GstRTSPUrl *) url);
107 result->media = media;
109 /* prealloc the streams now, filled with NULL */
110 n_streams = gst_rtsp_media_n_streams (media);
111 result->transports = g_ptr_array_new_full (n_streams, free_session_media);
112 g_ptr_array_set_size (result->transports, n_streams);
118 * gst_rtsp_session_media_set_transport:
119 * @media: a #GstRTSPSessionMedia
120 * @stream: a #GstRTSPStream
121 * @tr: a #GstRTSPTransport
123 * Configure the transport for @stream to @tr in @media.
125 * Returns: (transfer none): the new or updated #GstRTSPStreamTransport for @stream.
127 GstRTSPStreamTransport *
128 gst_rtsp_session_media_set_transport (GstRTSPSessionMedia * media,
129 GstRTSPStream * stream, GstRTSPTransport * tr)
131 GstRTSPStreamTransport *result;
133 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
134 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
135 g_return_val_if_fail (stream->idx < media->transports->len, NULL);
137 g_mutex_lock (&media->lock);
138 result = g_ptr_array_index (media->transports, stream->idx);
139 if (result == NULL) {
140 result = gst_rtsp_stream_transport_new (stream, tr);
141 g_ptr_array_index (media->transports, stream->idx) = result;
142 g_mutex_unlock (&media->lock);
144 gst_rtsp_stream_transport_set_transport (result, tr);
145 g_mutex_unlock (&media->lock);
152 * gst_rtsp_session_media_get_transport:
153 * @media: a #GstRTSPSessionMedia
154 * @idx: the stream index
156 * Get a previously created #GstRTSPStreamTransport for the stream at @idx.
158 * Returns: (transfer none): a #GstRTSPStreamTransport that is valid until the
159 * session of @media is unreffed.
161 GstRTSPStreamTransport *
162 gst_rtsp_session_media_get_transport (GstRTSPSessionMedia * media, guint idx)
164 GstRTSPStreamTransport *result;
166 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), NULL);
167 g_return_val_if_fail (idx < media->transports->len, NULL);
169 g_mutex_lock (&media->lock);
170 result = g_ptr_array_index (media->transports, idx);
171 g_mutex_unlock (&media->lock);
177 * gst_rtsp_session_media_alloc_channels:
178 * @media: a #GstRTSPSessionMedia
179 * @range: a #GstRTSPRange
181 * Fill @range with the next available min and max channels for
182 * interleaved transport.
184 * Returns: %TRUE on success.
187 gst_rtsp_session_media_alloc_channels (GstRTSPSessionMedia * media,
188 GstRTSPRange * range)
190 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
192 g_mutex_lock (&media->lock);
193 range->min = media->counter++;
194 range->max = media->counter++;
195 g_mutex_unlock (&media->lock);
201 * gst_rtsp_session_media_set_state:
202 * @media: a #GstRTSPSessionMedia
203 * @state: the new state
205 * Tell the media object @media to change to @state.
207 * Returns: %TRUE on success.
210 gst_rtsp_session_media_set_state (GstRTSPSessionMedia * media, GstState state)
214 g_return_val_if_fail (GST_IS_RTSP_SESSION_MEDIA (media), FALSE);
216 g_mutex_lock (&media->lock);
217 ret = gst_rtsp_media_set_state (media->media, state, media->transports);
218 g_mutex_unlock (&media->lock);