*
* 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__
typedef struct _GstRTSPMedia GstRTSPMedia;
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
-
-#include "rtsp-stream.h"
-#include "rtsp-auth.h"
+typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate;
/**
* GstRTSPMediaStatus:
* 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
*
* The state of the media pipeline.
GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
GST_RTSP_MEDIA_STATUS_PREPARING = 2,
GST_RTSP_MEDIA_STATUS_PREPARED = 3,
- GST_RTSP_MEDIA_STATUS_ERROR = 4
+ GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
+ GST_RTSP_MEDIA_STATUS_ERROR = 5
} GstRTSPMediaStatus;
/**
+ * 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 suspend mode of the media pipeline. A media pipeline is suspended right
+ * after creating the SDP and when the client performs a PAUSED request.
+ */
+typedef enum {
+ GST_RTSP_SUSPEND_MODE_NONE = 0,
+ GST_RTSP_SUSPEND_MODE_PAUSE = 1,
+ GST_RTSP_SUSPEND_MODE_RESET = 2
+} GstRTSPSuspendMode;
+
+/**
+ * GstRTSPTransportMode:
+ * @GST_RTSP_TRANSPORT_MODE_PLAY: Transport supports PLAY mode
+ * @GST_RTSP_TRANSPORT_MODE_RECORD: Transport supports RECORD mode
+ *
+ * The supported modes of the media.
+ */
+typedef enum {
+ 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
- * @protocols: the allowed lower transport for this stream
- * @reused: if this media has been reused
- * @is_ipv6: if this media is using ipv6
- * @element: the data providing element
- * @streams: the different #GstRTSPStream provided by @element
- * @dynamic: list of dynamic elements managed by @element
- * @status: the status of the media pipeline
- * @n_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
- * @seekable: if the pipeline can perform a seek
- * @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
* #GstRTSPStream objects that can produce data.
struct _GstRTSPMedia {
GObject parent;
- GMutex lock;
- GCond cond;
-
- gboolean shared;
- gboolean reusable;
- GstRTSPLowerTrans protocols;
- gboolean reused;
- gboolean is_ipv6;
- gboolean eos_shutdown;
- guint buffer_size;
- GstRTSPAuth *auth;
- gchar *multicast_group;
- guint mtu;
-
- GstElement *element;
- GPtrArray *streams;
- GList *dynamic;
- GstRTSPMediaStatus status;
- gint n_active;
- gboolean adding;
-
- /* the pipeline for the media */
- GstElement *pipeline;
- GstElement *fakesink;
- GSource *source;
- guint id;
-
- gboolean is_live;
- gboolean seekable;
- 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.
- * @handle_mtu: handle a mtu
+ * 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 (*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 (*prepared) (GstRTSPMedia *media);
- 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);
- gboolean (*new_state) (GstRTSPMedia *media, GstState state);
+ 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_auth (GstRTSPMedia *media, GstRTSPAuth *auth);
-GstRTSPAuth * gst_rtsp_media_get_auth (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_multicast_group (GstRTSPMedia *media, const gchar * mc);
-gchar * gst_rtsp_media_get_multicast_group (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_mtu (GstRTSPMedia *media, guint mtu);
-guint gst_rtsp_media_get_mtu (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);
+void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
+GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
+
+gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
+gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
+
+gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
+ GstSDPInfo * info);
+
+gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
+
+
/* creating streams */
void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
GstElement *payloader,
- GstPad *srcpad);
+ 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);
+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