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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
21 #include <gst/rtsp/gstrtsprange.h>
22 #include <gst/rtsp/gstrtspurl.h>
24 #ifndef __GST_RTSP_MEDIA_H__
25 #define __GST_RTSP_MEDIA_H__
29 /* types for the media */
30 #define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
31 #define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
32 #define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
33 #define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
34 #define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
35 #define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
36 #define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
37 #define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
39 typedef struct _GstRTSPMediaStream GstRTSPMediaStream;
40 typedef struct _GstRTSPMedia GstRTSPMedia;
41 typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
42 typedef struct _GstRTSPMediaTrans GstRTSPMediaTrans;
44 typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
45 typedef gboolean (*GstRTSPSendListFunc) (GstBufferList *blist, guint8 channel, gpointer user_data);
46 typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
50 * @idx: a stream index
51 * @send_rtp: callback for sending RTP messages
52 * @send_rtcp: callback for sending RTCP messages
53 * @user_data: user data passed in the callbacks
54 * @notify: free function for the user_data.
55 * @keep_alive: keep alive callback
56 * @ka_user_data: data passed to @keep_alive
57 * @ka_notify: called when @ka_user_data is freed
58 * @active: if we are actively sending
59 * @timeout: if we timed out
60 * @transport: a transport description
61 * @rtpsource: the receiver rtp source object
63 * A Transport description for stream @idx
65 struct _GstRTSPMediaTrans {
68 GstRTSPSendFunc send_rtp;
69 GstRTSPSendFunc send_rtcp;
70 GstRTSPSendListFunc send_rtp_list;
71 GstRTSPSendListFunc send_rtcp_list;
73 GDestroyNotify notify;
75 GstRTSPKeepAliveFunc keep_alive;
76 gpointer ka_user_data;
77 GDestroyNotify ka_notify;
81 GstRTSPTransport *transport;
88 * @srcpad: the srcpad of the stream
89 * @payloader: the payloader of the format
90 * @prepared: if the stream is prepared for streaming
91 * @server_port: the server udp ports
92 * @recv_rtp_sink: sinkpad for RTP buffers
93 * @recv_rtcp_sink: sinkpad for RTCP buffers
94 * @recv_rtp_src: srcpad for RTP buffers
95 * @recv_rtcp_src: srcpad for RTCP buffers
96 * @udpsrc: the udp source elements for RTP/RTCP
97 * @udpsink: the udp sink elements for RTP/RTCP
98 * @appsrc: the app source elements for RTP/RTCP
99 * @appsink: the app sink elements for RTP/RTCP
100 * @server_port: the server ports for this stream
101 * @caps_sig: the signal id for detecting caps
102 * @caps: the caps of the stream
103 * @tranports: the current transports being streamed
105 * The definition of a media stream. The streams are identified by @id.
107 struct _GstRTSPMediaStream {
109 GstElement *payloader;
112 /* pads on the rtpbin */
113 GstPad *recv_rtcp_sink;
114 GstPad *recv_rtp_sink;
115 GstPad *send_rtp_sink;
116 GstPad *send_rtp_src;
117 GstPad *send_rtcp_src;
119 /* the RTPSession object */
122 /* sinks used for sending and receiving RTP and RTCP, they share
124 GstElement *udpsrc[2];
125 GstElement *udpsink[2];
126 /* for TCP transport */
127 GstElement *appsrc[2];
128 GstElement *appsink[2];
131 GstElement *selector[2];
133 /* server ports for sending/receiving */
134 GstRTSPRange server_port;
136 /* the caps of the stream */
140 /* transports we stream to */
143 /* to filter out duplicate destinations in case multiudpsink is too old to do
145 gboolean filter_duplicates;
150 * GstRTSPMediaStatus:
151 * @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
152 * @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
153 * @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
154 * @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
156 * The state of the media pipeline.
159 GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
160 GST_RTSP_MEDIA_STATUS_PREPARING = 1,
161 GST_RTSP_MEDIA_STATUS_PREPARED = 2,
162 GST_RTSP_MEDIA_STATUS_ERROR = 3
163 } GstRTSPMediaStatus;
167 * @lock: for protecting the object
168 * @cond: for signaling the object
169 * @shared: if this media can be shared between clients
170 * @reusable: if this media can be reused after an unprepare
171 * @protocols: the allowed lower transport for this stream
172 * @reused: if this media has been reused
173 * @is_ipv6: if this media is using ipv6
174 * @element: the data providing element
175 * @streams: the different streams provided by @element
176 * @dynamic: list of dynamic elements managed by @element
177 * @status: the status of the media pipeline
178 * @active: the number of active connections
179 * @pipeline: the toplevel pipeline
180 * @fakesink: for making state changes async
181 * @source: the bus watch for pipeline messages.
182 * @id: the id of the watch
183 * @is_live: if the pipeline is live
184 * @buffering: if the pipeline is buffering
185 * @target_state: the desired target state of the pipeline
186 * @rtpbin: the rtpbin
187 * @range: the range of the media being streamed
189 * A class that contains the GStreamer element along with a list of
190 * #GstRTSPMediaStream objects that can produce data.
192 * This object is usually created from a #GstRTSPMediaFactory.
194 struct _GstRTSPMedia {
202 GstRTSPLowerTrans protocols;
205 gboolean eos_shutdown;
210 GstRTSPMediaStatus status;
212 gboolean eos_pending;
215 /* the pipeline for the media */
216 GstElement *pipeline;
217 GstElement *fakesink;
223 GstState target_state;
225 /* RTP session manager */
228 /* the range of media */
229 GstRTSPTimeRange range;
234 * @context: the main context for dispatching messages
235 * @loop: the mainloop for message.
236 * @thread: the thread dispatching messages.
237 * @handle_message: handle a message
238 * @unprepare: the default implementation sets the pipeline's state
241 * The RTSP media class
243 struct _GstRTSPMediaClass {
244 GObjectClass parent_class;
246 /* thread for the mainloop */
247 GMainContext *context;
252 gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
253 gboolean (*unprepare) (GstRTSPMedia *media);
256 gboolean (*prepared) (GstRTSPMedia *media);
257 gboolean (*unprepared) (GstRTSPMedia *media);
259 gboolean (*new_state) (GstRTSPMedia *media, GstState state);
262 GType gst_rtsp_media_get_type (void);
264 /* creating the media */
265 GstRTSPMedia * gst_rtsp_media_new (void);
267 void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
268 gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
270 void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
271 gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
273 void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
274 GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
276 void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
277 gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
279 /* prepare the media for playback */
280 gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
281 gboolean gst_rtsp_media_is_prepared (GstRTSPMedia *media);
282 gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
284 /* dealing with the media */
285 guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
286 GstRTSPMediaStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
288 gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
289 gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media, gboolean play);
291 GstFlowReturn gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer);
292 GstFlowReturn gst_rtsp_media_stream_rtcp (GstRTSPMediaStream *stream, GstBuffer *buffer);
294 gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transports);
296 void gst_rtsp_media_remove_elements (GstRTSPMedia *media);
298 void gst_rtsp_media_trans_cleanup (GstRTSPMediaTrans *trans);
302 #endif /* __GST_RTSP_MEDIA_H__ */