mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 13:41:48 +00:00
2498457b2f
Signals are not supposed to take ownership of their arguments but only borrow them for the scope of the signal emission. The old action signal `push-backchannel-buffer` is now marked deprecated. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3363>
349 lines
10 KiB
C
349 lines
10 KiB
C
/* 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;
|
|
GstClockTime base_time;
|
|
|
|
/* 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;
|
|
|
|
/* 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__ */
|