gstreamer/subprojects/gst-rtsp-server/gst/rtsp-sink/gstrtspclientsink.h

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/* GStreamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* <2006> Wim Taymans <wim@fluendo.com>
* <2015> Jan Schmidt <jan at centricular dot 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_RTSP_CLIENT_SINK_H__
#define __GST_RTSP_CLIENT_SINK_H__
#include <gst/gst.h>
G_BEGIN_DECLS
#include <gst/rtsp-server/rtsp-stream.h>
#include <gst/rtsp/rtsp.h>
#include <gio/gio.h>
#define GST_TYPE_RTSP_CLIENT_SINK \
(gst_rtsp_client_sink_get_type())
#define GST_RTSP_CLIENT_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSink))
#define GST_RTSP_CLIENT_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSinkClass))
#define GST_IS_RTSP_CLIENT_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSP_CLIENT_SINK))
#define GST_IS_RTSP_CLIENT_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSP_CLIENT_SINK))
#define GST_RTSP_CLIENT_SINK_CAST(obj) \
((GstRTSPClientSink *)(obj))
typedef struct _GstRTSPClientSink GstRTSPClientSink;
typedef struct _GstRTSPClientSinkClass GstRTSPClientSinkClass;
#define GST_RTSP_STATE_GET_LOCK(rtsp) (&GST_RTSP_CLIENT_SINK_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_RTSP_CLIENT_SINK_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 _GstRTSPStreamInfo GstRTSPStreamInfo;
typedef struct _GstRTSPStreamContext GstRTSPStreamContext;
struct _GstRTSPStreamContext {
GstRTSPClientSink *parent;
guint index;
/* Index of the SDPMedia in the stored SDP */
guint sdp_index;
GstElement *payloader;
guint payloader_block_id;
gboolean prerolled;
/* Stream management object */
GstRTSPStream *stream;
gboolean joined;
/* Secure profile key mgmt */
GstCaps *srtcpparams;
/* per stream connection */
GstRTSPConnInfo conninfo;
/* For interleaved mode */
guint8 channel[2];
GstRTSPStreamTransport *stream_transport;
guint ulpfec_percentage;
};
/**
* 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 _GstRTSPClientSink {
GstBin parent;
/* task and mutex for interleaved mode */
gboolean interleaved;
GstTask *task;
GRecMutex stream_rec_lock;
GstSegment segment;
gint free_channel;
/* UDP mode loop */
gint pending_cmd;
gint busy_cmd;
gboolean ignore_timeout;
gboolean open_error;
/* mutex for protecting state changes */
GRecMutex state_rec_lock;
GstSDPMessage *uri_sdp;
gboolean from_sdp;
/* properties */
GstRTSPLowerTrans protocols;
gboolean debug;
guint retry;
guint64 udp_timeout;
gint64 tcp_timeout;
guint latency;
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;
GstRTSPRange client_port_range;
gint udp_buffer_size;
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;
gint ntp_time_source;
gchar *user_agent;
GstRTSPPublishClockMode publish_clock_mode;
/* state */
GstRTSPState state;
gchar *content_base;
GstRTSPLowerTrans cur_protocols;
gboolean tried_url_auth;
gchar *addr;
GstRTSPAddressPool *pool;
gboolean need_redirect;
GstRTSPTimeRange *range;
gchar *control;
guint next_port_num;
GstClock *provided_clock;
/* supported methods */
gint methods;
/* session management */
GstRTSPConnInfo conninfo;
/* Everything goes in an internal
* locked-state bin */
GstBin *internal_bin;
/* Set to true when internal bin state
* >= PAUSED */
gboolean prerolled;
/* TRUE if we posted async-start */
gboolean in_async;
/* TRUE when stream info has been collected */
gboolean streams_collected;
/* TRUE when streams have been blocked */
guint n_streams_blocked;
GMutex block_streams_lock;
GCond block_streams_cond;
guint next_pad_id;
gint next_dyn_pt;
GstElement *rtpbin;
GList *contexts;
GstSDPMessage cursdp;
GMutex send_lock;
GMutex preroll_lock;
GCond preroll_cond;
/* TRUE if connection to server has been scheduled */
gboolean open_conn_start;
GMutex open_conn_lock;
GCond open_conn_cond;
GstClockTime rtx_time;
GstRTSPProfile profiles;
gchar *server_ip;
};
struct _GstRTSPClientSinkClass {
GstBinClass parent_class;
};
GType gst_rtsp_client_sink_get_type(void);
G_END_DECLS
#endif /* __GST_RTSP_CLIENT_SINK_H__ */