/* GStreamer * Copyright (C) <1999> Erik Walthinsen * <2006> 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. */ /* * Unless otherwise indicated, Source Code is licensed under MIT license. * See further explanation attached in License Statement (distributed in the file * LICENSE). * * Permission is hereby granted, free of charge, to any person obtaining a copy of * this software and associated documentation files (the "Software"), to deal in * the Software without restriction, including without limitation the rights to * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies * of the Software, and to permit persons to whom the Software is furnished to do * so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in all * copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE * SOFTWARE. */ #ifndef __GST_RTSPSRC_H__ #define __GST_RTSPSRC_H__ #include G_BEGIN_DECLS #include #include #include "gstrtspext.h" #define GST_TYPE_RTSPSRC \ (gst_rtspsrc_get_type()) #define GST_RTSPSRC(obj) \ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSPSRC,GstRTSPSrc)) #define GST_RTSPSRC_CLASS(klass) \ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSPSRC,GstRTSPSrcClass)) #define GST_IS_RTSPSRC(obj) \ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSPSRC)) #define GST_IS_RTSPSRC_CLASS(klass) \ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSPSRC)) #define GST_RTSPSRC_CAST(obj) \ ((GstRTSPSrc *)(obj)) typedef struct _GstRTSPSrc GstRTSPSrc; typedef struct _GstRTSPSrcClass GstRTSPSrcClass; #define GST_RTSP_STATE_GET_LOCK(rtsp) (&GST_RTSPSRC_CAST(rtsp)->state_rec_lock) #define GST_RTSP_STATE_LOCK(rtsp) (g_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp))) #define GST_RTSP_STATE_UNLOCK(rtsp) (g_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp))) #define GST_RTSP_STREAM_GET_LOCK(rtsp) (&GST_RTSPSRC_CAST(rtsp)->stream_rec_lock) #define GST_RTSP_STREAM_LOCK(rtsp) (g_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp))) #define GST_RTSP_STREAM_UNLOCK(rtsp) (g_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp))) typedef struct _GstRTSPConnInfo GstRTSPConnInfo; struct _GstRTSPConnInfo { gchar *location; GstRTSPUrl *url; gchar *url_str; GstRTSPConnection *connection; gboolean connected; gboolean flushing; GMutex send_lock; GMutex recv_lock; }; typedef struct _GstRTSPStream GstRTSPStream; struct _GstRTSPStream { gint id; GstRTSPSrc *parent; /* parent, no extra ref to parent is taken */ /* pad we expose or NULL when it does not have an actual pad */ GstPad *srcpad; GstFlowReturn last_ret; gboolean added; gboolean setup; gboolean skipped; gboolean eos; gboolean discont; gboolean need_caps; gboolean waiting_setup_response; /* for interleaved mode */ guint8 channel[2]; GstPad *channelpad[2]; /* our udp sources */ GstElement *udpsrc[2]; GstPad *blockedpad; gulong blockid; gboolean is_ipv6; /* our udp sinks back to the server */ GstElement *udpsink[2]; GstPad *rtcppad; /* fakesrc for sending dummy data or appsrc for sending backchannel data */ GstElement *rtpsrc; /* state */ guint port; gboolean container; gboolean is_real; guint8 default_pt; GstRTSPProfile profile; GArray *ptmap; /* original control url */ gchar *control_url; guint32 ssrc; guint32 seqbase; guint64 timebase; GstElement *srtpdec; GstCaps *srtcpparams; GstElement *srtpenc; guint32 send_ssrc; /* per stream connection */ GstRTSPConnInfo conninfo; /* session */ GObject *session; /* srtp key management */ GstMIKEYMessage *mikey; /* bandwidth */ guint as_bandwidth; guint rs_bandwidth; guint rr_bandwidth; /* destination */ gchar *destination; gboolean is_multicast; guint ttl; gboolean is_backchannel; /* A unique and stable id we will use for the stream start event */ gchar *stream_id; GstStructure *rtx_pt_map; guint32 segment_seqnum[2]; }; /** * GstRTSPSrcTimeoutCause: * @GST_RTSP_SRC_TIMEOUT_CAUSE_RTCP: timeout triggered by RTCP * * Different causes to why the rtspsrc generated the GstRTSPSrcTimeout * message. */ typedef enum { GST_RTSP_SRC_TIMEOUT_CAUSE_RTCP } GstRTSPSrcTimeoutCause; /** * GstRTSPNatMethod: * @GST_RTSP_NAT_NONE: none * @GST_RTSP_NAT_DUMMY: send dummy packets * * Different methods for trying to traverse firewalls. */ typedef enum { GST_RTSP_NAT_NONE, GST_RTSP_NAT_DUMMY } GstRTSPNatMethod; struct _GstRTSPSrc { GstBin parent; /* task and mutex for interleaved mode */ gboolean interleaved; GstTask *task; GRecMutex stream_rec_lock; GstSegment segment; gboolean running; gboolean need_range; gboolean server_side_trickmode; GstClockTime trickmode_interval; gint free_channel; gboolean need_segment; gboolean clip_out_segment; GstSegment out_segment; /* UDP mode loop */ gint pending_cmd; gint busy_cmd; GCond cmd_cond; gboolean ignore_timeout; gboolean open_error; /* mutex for protecting state changes */ GRecMutex state_rec_lock; GstSDPMessage *sdp; gboolean from_sdp; GList *streams; GstStructure *props; gboolean need_activate; /* properties */ GstRTSPLowerTrans protocols; gboolean debug; guint retry; guint64 udp_timeout; gint64 tcp_timeout; guint latency; gboolean drop_on_latency; guint64 connection_speed; GstRTSPNatMethod nat_method; gboolean do_rtcp; gboolean do_rtsp_keep_alive; gchar *proxy_host; guint proxy_port; gchar *proxy_user; /* from url or property */ gchar *proxy_passwd; /* from url or property */ gchar *prop_proxy_id; /* set via property */ gchar *prop_proxy_pw; /* set via property */ guint rtp_blocksize; gchar *user_id; gchar *user_pw; gint buffer_mode; GstRTSPRange client_port_range; gint udp_buffer_size; gboolean short_header; guint probation; gboolean udp_reconnect; gchar *multi_iface; gboolean ntp_sync; gboolean use_pipeline_clock; GstStructure *sdes; GTlsCertificateFlags tls_validation_flags; GTlsDatabase *tls_database; GTlsInteraction *tls_interaction; gboolean do_retransmission; gint ntp_time_source; gchar *user_agent; gint max_rtcp_rtp_time_diff; gboolean rfc7273_sync; gboolean add_reference_timestamp_meta; guint64 max_ts_offset_adjustment; gint64 max_ts_offset; gboolean max_ts_offset_is_set; gint backchannel; GstClockTime teardown_timeout; gboolean onvif_mode; gboolean onvif_rate_control; gboolean is_live; gboolean ignore_x_server_reply; GstStructure *prop_extra_http_request_headers; gboolean tcp_timestamp; gboolean force_non_compliant_url; /* state */ GstRTSPState state; gchar *content_base; GstRTSPLowerTrans cur_protocols; gboolean tried_url_auth; gchar *addr; gboolean need_redirect; GstRTSPTimeRange *range; gchar *control; guint next_port_num; GstClock *provided_clock; /* supported methods */ gint methods; /* seekability * -1.0 : Stream is not seekable * 0.0 : seekable only to the beginning * G_MAXFLOAT : Any value is possible * * Any other positive value indicates the longest duration * between any two random access points * */ gfloat seekable; guint32 seek_seqnum; GstClockTime last_pos; /* session management */ GstElement *manager; gulong manager_sig_id; gulong manager_ptmap_id; gboolean use_buffering; GstRTSPConnInfo conninfo; /* SET/GET PARAMETER requests queue */ GQueue set_get_param_q; /* a list of RTSP extensions as GstElement */ GstRTSPExtensionList *extensions; GstRTSPVersion default_version; GstRTSPVersion version; GstEvent *initial_seek; guint group_id; GMutex group_lock; }; struct _GstRTSPSrcClass { GstBinClass parent_class; /* action signals */ gboolean (*get_parameter) (GstRTSPSrc *rtsp, const gchar *parameter, const gchar *content_type, GstPromise *promise); gboolean (*get_parameters) (GstRTSPSrc *rtsp, gchar **parameters, const gchar *content_type, GstPromise *promise); gboolean (*set_parameter) (GstRTSPSrc *rtsp, const gchar *name, const gchar *value, const gchar *content_type, GstPromise *promise); GstFlowReturn (*push_backchannel_buffer) (GstRTSPSrc *src, guint id, GstSample *sample); GstFlowReturn (*push_backchannel_sample) (GstRTSPSrc *src, guint id, GstSample *sample); }; GType gst_rtspsrc_get_type(void); G_END_DECLS #endif /* __GST_RTSPSRC_H__ */