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gstreamer/gst/rtsp/gstrtspsrc.h
Mathieu Duponchelle 37eca8a12c rtspsrc: clip output segment on accurate seeks 
The output segment is only used in ONVIF mode.

The previous behaviour was to output a segment computed from
the Range response sent by the server.

In ONVIF mode, servers will start serving from the appropriate
synchronization point (keyframe), and the Range in response will
start at that position.

This means rtspsrc can now perform truly accurate seeks in that
mode, by clipping the output segment to the values requested in
the seek. The decoder will then discard out of segment buffers
and playback will start without artefacts at the exact requested
position, similar to the behaviour of a demuxer when an accurate
seek is requested.
2019-08-30 14:50:21 +00:00

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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;
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;
GTimeVal tcp_timeout;
GTimeVal *ptcp_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;
GstClockTime max_rtcp_rtp_time_diff;
gboolean rfc7273_sync;
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;
/* 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;
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;
};
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);
};
GType gst_rtspsrc_get_type(void);
G_END_DECLS
#endif /* __GST_RTSPSRC_H__ */
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