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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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87c177567d
Adds an address pool for rtspclientsink in order to allow the "port-range" property to restrict the ports available for the RTSP streams rather than always using the ephemeral port-range. If a value is not provided to the "port-range" property, rtspclientsink will select random ports from the ephemeral port-range as before. Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2606 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4828>
261 lines
8 KiB
C
261 lines
8 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* <2006> Wim Taymans <wim@fluendo.com>
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* <2015> Jan Schmidt <jan at centricular dot com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/*
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* Unless otherwise indicated, Source Code is licensed under MIT license.
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* See further explanation attached in License Statement (distributed in the file
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* LICENSE).
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*
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* Permission is hereby granted, free of charge, to any person obtaining a copy of
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* this software and associated documentation files (the "Software"), to deal in
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* the Software without restriction, including without limitation the rights to
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* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
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* of the Software, and to permit persons to whom the Software is furnished to do
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* so, subject to the following conditions:
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*
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* The above copyright notice and this permission notice shall be included in all
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* copies or substantial portions of the Software.
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*
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* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
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* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
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* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
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* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
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* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
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* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
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* SOFTWARE.
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*/
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#ifndef __GST_RTSP_CLIENT_SINK_H__
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#define __GST_RTSP_CLIENT_SINK_H__
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#include <gst/gst.h>
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G_BEGIN_DECLS
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#include <gst/rtsp-server/rtsp-stream.h>
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#include <gst/rtsp/rtsp.h>
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#include <gio/gio.h>
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#define GST_TYPE_RTSP_CLIENT_SINK \
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(gst_rtsp_client_sink_get_type())
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#define GST_RTSP_CLIENT_SINK(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSink))
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#define GST_RTSP_CLIENT_SINK_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSP_CLIENT_SINK,GstRTSPClientSinkClass))
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#define GST_IS_RTSP_CLIENT_SINK(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSP_CLIENT_SINK))
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#define GST_IS_RTSP_CLIENT_SINK_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSP_CLIENT_SINK))
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#define GST_RTSP_CLIENT_SINK_CAST(obj) \
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((GstRTSPClientSink *)(obj))
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typedef struct _GstRTSPClientSink GstRTSPClientSink;
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typedef struct _GstRTSPClientSinkClass GstRTSPClientSinkClass;
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#define GST_RTSP_STATE_GET_LOCK(rtsp) (&GST_RTSP_CLIENT_SINK_CAST(rtsp)->state_rec_lock)
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#define GST_RTSP_STATE_LOCK(rtsp) (g_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
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#define GST_RTSP_STATE_UNLOCK(rtsp) (g_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
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#define GST_RTSP_STREAM_GET_LOCK(rtsp) (&GST_RTSP_CLIENT_SINK_CAST(rtsp)->stream_rec_lock)
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#define GST_RTSP_STREAM_LOCK(rtsp) (g_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
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#define GST_RTSP_STREAM_UNLOCK(rtsp) (g_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
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typedef struct _GstRTSPConnInfo GstRTSPConnInfo;
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struct _GstRTSPConnInfo {
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gchar *location;
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GstRTSPUrl *url;
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gchar *url_str;
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GstRTSPConnection *connection;
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gboolean connected;
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gboolean flushing;
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GMutex send_lock;
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GMutex recv_lock;
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};
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typedef struct _GstRTSPStreamInfo GstRTSPStreamInfo;
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typedef struct _GstRTSPStreamContext GstRTSPStreamContext;
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struct _GstRTSPStreamContext {
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GstRTSPClientSink *parent;
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guint index;
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/* Index of the SDPMedia in the stored SDP */
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guint sdp_index;
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GstElement *payloader;
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guint payloader_block_id;
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gboolean prerolled;
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/* Stream management object */
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GstRTSPStream *stream;
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gboolean joined;
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/* Secure profile key mgmt */
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GstCaps *srtcpparams;
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/* per stream connection */
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GstRTSPConnInfo conninfo;
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/* For interleaved mode */
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guint8 channel[2];
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GstRTSPStreamTransport *stream_transport;
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guint ulpfec_percentage;
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};
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/**
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* GstRTSPNatMethod:
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* @GST_RTSP_NAT_NONE: none
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* @GST_RTSP_NAT_DUMMY: send dummy packets
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*
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* Different methods for trying to traverse firewalls.
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*/
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typedef enum
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{
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GST_RTSP_NAT_NONE,
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GST_RTSP_NAT_DUMMY
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} GstRTSPNatMethod;
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struct _GstRTSPClientSink {
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GstBin parent;
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/* task and mutex for interleaved mode */
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gboolean interleaved;
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GstTask *task;
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GRecMutex stream_rec_lock;
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GstSegment segment;
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gint free_channel;
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/* UDP mode loop */
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gint pending_cmd;
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gint busy_cmd;
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gboolean ignore_timeout;
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gboolean open_error;
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/* mutex for protecting state changes */
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GRecMutex state_rec_lock;
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GstSDPMessage *uri_sdp;
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gboolean from_sdp;
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/* properties */
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GstRTSPLowerTrans protocols;
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gboolean debug;
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guint retry;
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guint64 udp_timeout;
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gint64 tcp_timeout;
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guint latency;
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gboolean do_rtsp_keep_alive;
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gchar *proxy_host;
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guint proxy_port;
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gchar *proxy_user; /* from url or property */
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gchar *proxy_passwd; /* from url or property */
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gchar *prop_proxy_id; /* set via property */
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gchar *prop_proxy_pw; /* set via property */
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guint rtp_blocksize;
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gchar *user_id;
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gchar *user_pw;
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GstRTSPRange client_port_range;
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gint udp_buffer_size;
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gboolean udp_reconnect;
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gchar *multi_iface;
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gboolean ntp_sync;
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gboolean use_pipeline_clock;
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GstStructure *sdes;
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GTlsCertificateFlags tls_validation_flags;
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GTlsDatabase *tls_database;
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GTlsInteraction *tls_interaction;
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gint ntp_time_source;
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gchar *user_agent;
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GstRTSPPublishClockMode publish_clock_mode;
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/* state */
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GstRTSPState state;
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gchar *content_base;
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GstRTSPLowerTrans cur_protocols;
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gboolean tried_url_auth;
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gchar *addr;
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GstRTSPAddressPool *pool;
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gboolean need_redirect;
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GstRTSPTimeRange *range;
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gchar *control;
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guint next_port_num;
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GstClock *provided_clock;
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/* supported methods */
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gint methods;
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/* session management */
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GstRTSPConnInfo conninfo;
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/* Everything goes in an internal
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* locked-state bin */
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GstBin *internal_bin;
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/* Set to true when internal bin state
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* >= PAUSED */
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gboolean prerolled;
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/* TRUE if we posted async-start */
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gboolean in_async;
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/* TRUE when stream info has been collected */
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gboolean streams_collected;
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/* TRUE when streams have been blocked */
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guint n_streams_blocked;
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GMutex block_streams_lock;
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GCond block_streams_cond;
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guint next_pad_id;
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gint next_dyn_pt;
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GstElement *rtpbin;
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GList *contexts;
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GstSDPMessage cursdp;
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GMutex send_lock;
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GMutex preroll_lock;
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GCond preroll_cond;
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/* TRUE if connection to server has been scheduled */
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gboolean open_conn_start;
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GMutex open_conn_lock;
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GCond open_conn_cond;
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GstClockTime rtx_time;
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GstRTSPProfile profiles;
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gchar *server_ip;
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};
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struct _GstRTSPClientSinkClass {
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GstBinClass parent_class;
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};
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GType gst_rtsp_client_sink_get_type(void);
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G_END_DECLS
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#endif /* __GST_RTSP_CLIENT_SINK_H__ */
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