/* 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, 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 _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); /** * 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 * * A Transport description for stream @idx */ 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; }; /** * 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 * * The definition of a media stream. The streams are identified by @id. */ 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; /* to filter out duplicate destinations in case multiudpsink is too old to do * this for us */ gboolean filter_duplicates; GList *destinations; }; /** * 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 * * The state of the media pipeline. */ 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; /** * 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 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 * #GstRTSPMediaStream 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; GstElement *element; GArray *streams; GList *dynamic; GstRTSPMediaStatus status; gint active; gboolean eos_pending; gboolean adding; /* 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; }; /** * 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. * * 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 (*prepared) (GstRTSPMedia *media); gboolean (*unprepared) (GstRTSPMedia *media); }; 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); /* prepare the media for playback */ gboolean gst_rtsp_media_prepare (GstRTSPMedia *media); gboolean gst_rtsp_media_is_prepared (GstRTSPMedia *media); 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); gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range); GstFlowReturn gst_rtsp_media_stream_rtp (GstRTSPMediaStream *stream, GstBuffer *buffer); GstFlowReturn gst_rtsp_media_stream_rtcp (GstRTSPMediaStream *stream, GstBuffer *buffer); gboolean gst_rtsp_media_set_state (GstRTSPMedia *media, GstState state, GArray *transports); void gst_rtsp_media_remove_elements (GstRTSPMedia *media); void gst_rtsp_media_trans_cleanup (GstRTSPMediaTrans *trans); G_END_DECLS #endif /* __GST_RTSP_MEDIA_H__ */