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gstreamer/subprojects/gst-plugins-good/gst/rtsp/gstrtspsrc.h

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* <2006> Wim Taymans <wim@fluendo.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.
*/
/*
* 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 <gst/gst.h>
G_BEGIN_DECLS
#include <gst/rtsp/rtsp.h>
#include <gio/gio.h>
#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__ */
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