diff options
Diffstat (limited to 'include/gst/rtsp-server/rtsp-media.h')
-rw-r--r-- | include/gst/rtsp-server/rtsp-media.h | 449 |
1 files changed, 449 insertions, 0 deletions
diff --git a/include/gst/rtsp-server/rtsp-media.h b/include/gst/rtsp-server/rtsp-media.h new file mode 100644 index 0000000000..9c2494a64e --- /dev/null +++ b/include/gst/rtsp-server/rtsp-media.h @@ -0,0 +1,449 @@ +/* GStreamer + * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * 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., 51 Franklin St, Fifth Floor, + * Boston, MA 02110-1301, USA. + */ + +#include <gst/gst.h> +#include <gst/rtsp/rtsp.h> +#include <gst/net/gstnet.h> + +#ifndef __GST_RTSP_MEDIA_H__ +#define __GST_RTSP_MEDIA_H__ + +#include "rtsp-server-prelude.h" + +G_BEGIN_DECLS + +/* types for the media */ +#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ()) +#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA)) +#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA)) +#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass)) +#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia)) +#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass)) +#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj)) +#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass)) + +typedef struct _GstRTSPMedia GstRTSPMedia; +typedef struct _GstRTSPMediaClass GstRTSPMediaClass; +typedef struct _GstRTSPMediaPrivate GstRTSPMediaPrivate; + +/** + * 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 + * + * The state of the media pipeline. + */ +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; + +/** + * 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; + +/** + * GstRTSPPublishClockMode: + * @GST_RTSP_PUBLISH_CLOCK_MODE_NONE: Publish nothing + * @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK: Publish the clock but not the offset + * @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET: Publish the clock and offset + * + * Whether the clock and possibly RTP/clock offset should be published according to RFC7273. + */ +typedef enum { + GST_RTSP_PUBLISH_CLOCK_MODE_NONE, + GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK, + GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET +} GstRTSPPublishClockMode; + +#define GST_TYPE_RTSP_TRANSPORT_MODE (gst_rtsp_transport_mode_get_type()) +GST_RTSP_SERVER_API +GType gst_rtsp_transport_mode_get_type (void); + +#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type()) +GST_RTSP_SERVER_API +GType gst_rtsp_suspend_mode_get_type (void); + +#define GST_TYPE_RTSP_PUBLISH_CLOCK_MODE (gst_rtsp_publish_clock_mode_get_type()) +GST_RTSP_SERVER_API +GType gst_rtsp_publish_clock_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: + * + * A class that contains the GStreamer element along with a list of + * #GstRTSPStream objects that can produce data. + * + * This object is usually created from a #GstRTSPMediaFactory. + */ +struct _GstRTSPMedia { + GObject parent; + + /*< private >*/ + GstRTSPMediaPrivate *priv; + gpointer _gst_reserved[GST_PADDING]; +}; + +/** + * GstRTSPMediaClass: + * @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 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; + + /* vmethods */ + gboolean (*handle_message) (GstRTSPMedia *media, GstMessage *message); + gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread); + 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); + + /* signals */ + 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); + + /*< private >*/ + gpointer _gst_reserved[GST_PADDING_LARGE-1]; +}; + +GST_RTSP_SERVER_API +GType gst_rtsp_media_get_type (void); + +/* creating the media */ + +GST_RTSP_SERVER_API +GstRTSPMedia * gst_rtsp_media_new (GstElement *element); + +GST_RTSP_SERVER_API +GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline); + +GST_RTSP_SERVER_API +GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_permissions (GstRTSPMedia *media, + GstRTSPPermissions *permissions); + +GST_RTSP_SERVER_API +GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia *media, gboolean stop_on_disconnect); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_transport_mode (GstRTSPMedia *media, GstRTSPTransportMode mode); + +GST_RTSP_SERVER_API +GstRTSPTransportMode gst_rtsp_media_get_transport_mode (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles); + +GST_RTSP_SERVER_API +GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols); + +GST_RTSP_SERVER_API +GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool); + +GST_RTSP_SERVER_API +GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_multicast_iface (GstRTSPMedia *media, const gchar *multicast_iface); + +GST_RTSP_SERVER_API +gchar * gst_rtsp_media_get_multicast_iface (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size); + +GST_RTSP_SERVER_API +guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_retransmission_time (GstRTSPMedia *media, GstClockTime time); + +GST_RTSP_SERVER_API +GstClockTime gst_rtsp_media_get_retransmission_time (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media, + gboolean do_retransmission); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_latency (GstRTSPMedia *media, guint latency); + +GST_RTSP_SERVER_API +guint gst_rtsp_media_get_latency (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media, + const gchar *address, guint16 port); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_clock (GstRTSPMedia *media, GstClock * clock); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media, GstRTSPPublishClockMode mode); + +GST_RTSP_SERVER_API +GstRTSPPublishClockMode gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia *media, guint ttl); + +GST_RTSP_SERVER_API +guint gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia *media, gboolean bind_mcast_addr); +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_dscp_qos (GstRTSPMedia * media, gint dscp_qos); +GST_RTSP_SERVER_API +gint gst_rtsp_media_get_dscp_qos (GstRTSPMedia * media); + +/* prepare the media for playback */ + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode); + +GST_RTSP_SERVER_API +GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_suspend (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, + GstSDPInfo * info); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp); + +/* creating streams */ + +GST_RTSP_SERVER_API +void gst_rtsp_media_collect_streams (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media, + GstElement *payloader, + GstPad *pad); + +/* dealing with the media */ + +GST_RTSP_SERVER_API +void gst_rtsp_media_lock (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_unlock (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +guint gst_rtsp_media_n_streams (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx); + +GST_RTSP_SERVER_API +GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_seek_full (GstRTSPMedia *media, + GstRTSPTimeRange *range, + GstSeekFlags flags); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_seek_trickmode (GstRTSPMedia *media, + GstRTSPTimeRange *range, + GstSeekFlags flags, + gdouble rate, + GstClockTime trickmode_interval); + +GST_RTSP_SERVER_API +GstClockTimeDiff gst_rtsp_media_seekable (GstRTSPMedia *media); + +GST_RTSP_SERVER_API +gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media, + gboolean play, + GstRTSPRangeUnit unit); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_get_rates (GstRTSPMedia * media, + gdouble * rate, + gdouble * applied_rate); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, + GPtrArray *transports); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, + GstState state); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_is_receive_only (GstRTSPMedia * media); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_has_completed_sender (GstRTSPMedia * media); + +GST_RTSP_SERVER_API +void gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled); + +GST_RTSP_SERVER_API +gboolean gst_rtsp_media_get_rate_control (GstRTSPMedia * media); + +#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC +G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPMedia, gst_object_unref) +#endif + +G_END_DECLS + +#endif /* __GST_RTSP_MEDIA_H__ */ |