/* GStreamer * Copyright (C) 2008 Wim Taymans * * 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 #include #include #ifndef __GST_RTSP_MEDIA_H__ #define __GST_RTSP_MEDIA_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; #include "rtsp-stream.h" #include "rtsp-auth.h" #include "rtsp-address-pool.h" /** * 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_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_ERROR = 4 } GstRTSPMediaStatus; /** * GstRTSPMedia: * @parent: parent GObject * @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 * @eos_shutdown: if EOS should be sent on shutdown * @buffer_size: The UDP buffer size * @auth: the authentication service in use * @multicast_group: the multicast group to use * @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 * @adding: when elements are added to the pipeline * @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 * @range_start: range start in #GstClockTime * @range_stop: range stop in #GstClockTime * * 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; GMutex lock; GCond cond; gboolean shared; gboolean reusable; GstRTSPLowerTrans protocols; gboolean reused; gboolean is_ipv6; gboolean eos_shutdown; guint buffer_size; GstRTSPAuth *auth; GstRTSPAddressPool*pool; GstElement *element; GRecMutex state_lock; 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; GstClockTime range_start; GstClockTime range_stop; }; /** * GstRTSPMediaClass: * @context: the main context for dispatching messages * @loop: the mainloop for message. * @thread: the thread dispatching messages. * @handle_message: handle a message * @unprepare: the default implementation sets the pipeline's state * to GST_STATE_NULL and removes all elements. * * 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); /* signals */ gboolean (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream); gboolean (*prepared) (GstRTSPMedia *media); gboolean (*unprepared) (GstRTSPMedia *media); gboolean (*new_state) (GstRTSPMedia *media, GstState state); }; GType gst_rtsp_media_get_type (void); /* creating the media */ GstRTSPMedia * gst_rtsp_media_new (void); void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared); gboolean gst_rtsp_media_is_shared (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_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); /* prepare the media for playback */ gboolean gst_rtsp_media_prepare (GstRTSPMedia *media); gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media); /* creating streams */ void gst_rtsp_media_collect_streams (GstRTSPMedia *media); GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media, GstElement *payloader, GstPad *srcpad); /* dealing with the media */ guint gst_rtsp_media_n_streams (GstRTSPMedia *media); GstRTSPStream * gst_rtsp_media_get_stream (GstRTSPMedia *media, guint idx); gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range); gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media, gboolean play); gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GPtrArray *transports); G_END_DECLS #endif /* __GST_RTSP_MEDIA_H__ */