*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
*/
#include <gst/gst.h>
-#include <gst/rtsp/gstrtsprange.h>
-#include <gst/rtsp/gstrtspurl.h>
+#include <gst/rtsp/rtsp.h>
+#include <gst/net/gstnet.h>
#ifndef __GST_RTSP_MEDIA_H__
#define __GST_RTSP_MEDIA_H__
#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
-typedef struct _GstRTSPMediaStream GstRTSPMediaStream;
typedef struct _GstRTSPMedia GstRTSPMedia;
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
-typedef struct _GstRTSPMediaTrans GstRTSPMediaTrans;
-
-typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
-typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
+typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate;
/**
- * GstRTSPMediaTrans:
- * @idx: a stream index
- * @send_rtp: callback for sending RTP messages
- * @send_rtcp: callback for sending RTCP messages
- * @user_data: user data passed in the callbacks
- * @notify: free function for the user_data.
- * @keep_alive: keep alive callback
- * @ka_user_data: data passed to @keep_alive
- * @ka_notify: called when @ka_user_data is freed
- * @active: if we are actively sending
- * @timeout: if we timed out
- * @transport: a transport description
- * @rtpsource: the receiver rtp source object
+ * GstRTSPMediaStatus:
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
+ * @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
+ * shutdown.
+ * @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
+ * @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
+ * @GST_RTSP_MEDIA_STATUS_SUSPENDED: media is suspended
+ * @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
*
- * A Transport description for stream @idx
+ * The state of the media pipeline.
*/
-struct _GstRTSPMediaTrans {
- guint idx;
-
- GstRTSPSendFunc send_rtp;
- GstRTSPSendFunc send_rtcp;
- gpointer user_data;
- GDestroyNotify notify;
-
- GstRTSPKeepAliveFunc keep_alive;
- gpointer ka_user_data;
- GDestroyNotify ka_notify;
- gboolean active;
- gboolean timeout;
-
- GstRTSPTransport *transport;
-
- GObject *rtpsource;
-};
+typedef enum {
+ GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
+ GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
+ GST_RTSP_MEDIA_STATUS_PREPARING = 2,
+ GST_RTSP_MEDIA_STATUS_PREPARED = 3,
+ GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
+ GST_RTSP_MEDIA_STATUS_ERROR = 5
+} GstRTSPMediaStatus;
/**
- * GstRTSPMediaStream:
- *
- * @srcpad: the srcpad of the stream
- * @payloader: the payloader of the format
- * @prepared: if the stream is prepared for streaming
- * @server_port: the server udp ports
- * @recv_rtp_sink: sinkpad for RTP buffers
- * @recv_rtcp_sink: sinkpad for RTCP buffers
- * @recv_rtp_src: srcpad for RTP buffers
- * @recv_rtcp_src: srcpad for RTCP buffers
- * @udpsrc: the udp source elements for RTP/RTCP
- * @udpsink: the udp sink elements for RTP/RTCP
- * @appsrc: the app source elements for RTP/RTCP
- * @appsink: the app sink elements for RTP/RTCP
- * @server_port: the server ports for this stream
- * @caps_sig: the signal id for detecting caps
- * @caps: the caps of the stream
- * @tranports: the current transports being streamed
+ * GstRTSPSuspendMode:
+ * @GST_RTSP_SUSPEND_MODE_NONE: Media is not suspended
+ * @GST_RTSP_SUSPEND_MODE_PAUSE: Media is PAUSED in suspend
+ * @GST_RTSP_SUSPEND_MODE_RESET: The media is set to NULL when suspended
*
- * The definition of a media stream. The streams are identified by @id.
+ * The suspend mode of the media pipeline. A media pipeline is suspended right
+ * after creating the SDP and when the client performs a PAUSED request.
*/
-struct _GstRTSPMediaStream {
- GstPad *srcpad;
- GstElement *payloader;
- gboolean prepared;
-
- /* pads on the rtpbin */
- GstPad *recv_rtcp_sink;
- GstPad *recv_rtp_sink;
- GstPad *send_rtp_sink;
- GstPad *send_rtp_src;
- GstPad *send_rtcp_src;
-
- /* the RTPSession object */
- GObject *session;
-
- /* sinks used for sending and receiving RTP and RTCP, they share
- * sockets */
- GstElement *udpsrc[2];
- GstElement *udpsink[2];
- /* for TCP transport */
- GstElement *appsrc[2];
- GstElement *appsink[2];
-
- GstElement *tee[2];
- GstElement *selector[2];
-
- /* server ports for sending/receiving */
- GstRTSPRange server_port;
-
- /* the caps of the stream */
- gulong caps_sig;
- GstCaps *caps;
-
- /* transports we stream to */
- GList *transports;
-};
+typedef enum {
+ GST_RTSP_SUSPEND_MODE_NONE = 0,
+ GST_RTSP_SUSPEND_MODE_PAUSE = 1,
+ GST_RTSP_SUSPEND_MODE_RESET = 2
+} GstRTSPSuspendMode;
/**
- * GstRTSPMediaStatus:
- * @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
- * @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
- * @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
- * @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
+ * GstRTSPTransportMode:
+ * @GST_RTSP_TRANSPORT_MODE_PLAY: Transport supports PLAY mode
+ * @GST_RTSP_TRANSPORT_MODE_RECORD: Transport supports RECORD mode
*
- * The state of the media pipeline.
+ * The supported modes of the media.
*/
typedef enum {
- GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
- GST_RTSP_MEDIA_STATUS_PREPARING = 1,
- GST_RTSP_MEDIA_STATUS_PREPARED = 2,
- GST_RTSP_MEDIA_STATUS_ERROR = 3
-} GstRTSPMediaStatus;
+ GST_RTSP_TRANSPORT_MODE_PLAY = 1,
+ GST_RTSP_TRANSPORT_MODE_RECORD = 2,
+} GstRTSPTransportMode;
+
+#define GST_TYPE_RTSP_TRANSPORT_MODE (gst_rtsp_transport_mode_get_type())
+GType gst_rtsp_transport_mode_get_type (void);
+
+#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
+GType gst_rtsp_suspend_mode_get_type (void);
+
+#include "rtsp-stream.h"
+#include "rtsp-thread-pool.h"
+#include "rtsp-permissions.h"
+#include "rtsp-address-pool.h"
+#include "rtsp-sdp.h"
/**
* GstRTSPMedia:
- * @lock: for protecting the object
- * @cond: for signaling the object
- * @shared: if this media can be shared between clients
- * @reusable: if this media can be reused after an unprepare
- * @reused: if this media has been reused
- * @element: the data providing element
- * @streams: the different streams provided by @element
- * @dynamic: list of dynamic elements managed by @element
- * @status: the status of the media pipeline
- * @active: the number of active connections
- * @pipeline: the toplevel pipeline
- * @fakesink: for making state changes async
- * @source: the bus watch for pipeline messages.
- * @id: the id of the watch
- * @is_live: if the pipeline is live
- * @buffering: if the pipeline is buffering
- * @target_state: the desired target state of the pipeline
- * @rtpbin: the rtpbin
- * @range: the range of the media being streamed
*
* A class that contains the GStreamer element along with a list of
- * #GstRTSPediaStream objects that can produce data.
+ * #GstRTSPStream objects that can produce data.
*
* This object is usually created from a #GstRTSPMediaFactory.
*/
struct _GstRTSPMedia {
GObject parent;
- GMutex *lock;
- GCond *cond;
-
- gboolean shared;
- gboolean reusable;
- gboolean reused;
- gboolean is_ipv6;
-
- GstElement *element;
- GArray *streams;
- GList *dynamic;
- GstRTSPMediaStatus status;
- gint active;
-
- /* the pipeline for the media */
- GstElement *pipeline;
- GstElement *fakesink;
- GSource *source;
- guint id;
-
- gboolean is_live;
- gboolean buffering;
- GstState target_state;
-
- /* RTP session manager */
- GstElement *rtpbin;
-
- /* the range of media */
- GstRTSPTimeRange range;
+ /*< private >*/
+ GstRTSPMediaPrivate *priv;
+ gpointer _gst_reserved[GST_PADDING];
};
/**
* GstRTSPMediaClass:
- * @context: the main context for dispatching messages
- * @loop: the mainloop for message.
- * @thread: the thread dispatching messages.
* @handle_message: handle a message
+ * @prepare: the default implementation adds all elements and sets the
+ * pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
+ * in case of NO_PREROLL elements).
* @unprepare: the default implementation sets the pipeline's state
- * to GST_STATE_NULL.
+ * to GST_STATE_NULL and removes all elements.
+ * @suspend: the default implementation sets the pipeline's state to
+ * GST_STATE_NULL GST_STATE_PAUSED depending on the selected
+ * suspend mode.
+ * @unsuspend: the default implementation reverts the suspend operation.
+ * The pipeline will be prerolled again if it's state was
+ * set to GST_STATE_NULL in suspend.
+ * @convert_range: convert a range to the given unit
+ * @query_position: query the current position in the pipeline
+ * @query_stop: query when playback will stop
*
* The RTSP media class
*/
struct _GstRTSPMediaClass {
GObjectClass parent_class;
- /* thread for the mainloop */
- GMainContext *context;
- GMainLoop *loop;
- GThread *thread;
-
/* vmethods */
- gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
- gboolean (*unprepare) (GstRTSPMedia *media);
+ gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message);
+ gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread);
+ gboolean (*start_preroll) (GstRTSPMedia *media);
+ gboolean (*unprepare) (GstRTSPMedia *media);
+ gboolean (*suspend) (GstRTSPMedia *media);
+ gboolean (*unsuspend) (GstRTSPMedia *media);
+ gboolean (*convert_range) (GstRTSPMedia *media, GstRTSPTimeRange *range,
+ GstRTSPRangeUnit unit);
+ gboolean (*query_position) (GstRTSPMedia *media, gint64 *position);
+ gboolean (*query_stop) (GstRTSPMedia *media, gint64 *stop);
+ GstElement * (*create_rtpbin) (GstRTSPMedia *media);
+ gboolean (*setup_rtpbin) (GstRTSPMedia *media, GstElement *rtpbin);
+ gboolean (*setup_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
+ gboolean (*start_prepare) (GstRTSPMedia *media);
/* signals */
- gboolean (*unprepared) (GstRTSPMedia *media);
+ void (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+ void (*removed_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
+
+ void (*prepared) (GstRTSPMedia *media);
+ void (*unprepared) (GstRTSPMedia *media);
+
+ void (*target_state) (GstRTSPMedia *media, GstState state);
+ void (*new_state) (GstRTSPMedia *media, GstState state);
+
+ gboolean (*handle_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp);
+
+ void (*preparing) (GstRTSPMedia *media, GstRTSPStream * stream, guint idx);
+ void (*unpreparing) (GstRTSPMedia *media, GstRTSPStream * stream, guint idx);
+ /*< private >*/
+ gpointer _gst_reserved[GST_PADDING_LARGE-1];
};
GType gst_rtsp_media_get_type (void);
/* creating the media */
-GstRTSPMedia * gst_rtsp_media_new (void);
+GstRTSPMedia * gst_rtsp_media_new (GstElement *element);
+GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media);
+
+void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
+
+GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_permissions (GstRTSPMedia *media,
+ GstRTSPPermissions *permissions);
+GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
+void gst_rtsp_media_set_transport_mode (GstRTSPMedia *media, GstRTSPTransportMode mode);
+GstRTSPTransportMode gst_rtsp_media_get_transport_mode (GstRTSPMedia *media);
+
void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
+void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles);
+GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
+GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
+gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
+GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
+guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_retransmission_time (GstRTSPMedia *media, GstClockTime time);
+GstClockTime gst_rtsp_media_get_retransmission_time (GstRTSPMedia *media);
+
+void gst_rtsp_media_set_latency (GstRTSPMedia *media, guint latency);
+guint gst_rtsp_media_get_latency (GstRTSPMedia *media);
+
+void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
+gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
+GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
+ const gchar *address, guint16 port);
+
/* prepare the media for playback */
-gboolean gst_rtsp_media_prepare (GstRTSPMedia *media);
-gboolean gst_rtsp_media_is_prepared (GstRTSPMedia *media);
+gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread);
gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
-/* dealing with the media */
-guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
-GstRTSPMediaStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
+void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
+GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
-gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
+gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
+gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
-GstFlowReturn gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer);
-GstFlowReturn gst_rtsp_media_stream_rtcp (GstRTSPMediaStream *stream, GstBuffer *buffer);
+gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
-gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *trans);
+gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
-void gst_rtsp_media_remove_elements (GstRTSPMedia *media);
+
+/* creating streams */
+void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
+GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
+ GstElement *payloader,
+ GstPad *pad);
+
+/* dealing with the media */
+GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media);
+GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media);
+
+guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
+GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx);
+GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control);
+
+gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
+gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media,
+ gboolean play,
+ GstRTSPRangeUnit unit);
+
+gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state,
+ GPtrArray *transports);
+void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
+ GstState state);
+GstStateChangeReturn gst_rtsp_media_set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state);
+void gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status);
+
+GstElement * gst_rtsp_media_get_pipeline (GstRTSPMedia * media);
+GstElement * gst_rtsp_media_get_rtpbin (GstRTSPMedia * media);
G_END_DECLS
projects / platform / upstream / gst-rtsp-server.git / blobdiff