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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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ae84ae1b36
Allow setting an SDP with the rtsp-sdp:// url. Based on patch from Marco Ballesio. See #628214
252 lines
7.6 KiB
C
252 lines
7.6 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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*
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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_RTSPSRC_H__
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#define __GST_RTSPSRC_H__
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#include <gst/gst.h>
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G_BEGIN_DECLS
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#include <gst/rtsp/gstrtspconnection.h>
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#include <gst/rtsp/gstrtspmessage.h>
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#include <gst/rtsp/gstrtspurl.h>
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#include <gst/rtsp/gstrtsprange.h>
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#include "gstrtspext.h"
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#define GST_TYPE_RTSPSRC \
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(gst_rtspsrc_get_type())
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#define GST_RTSPSRC(obj) \
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(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSPSRC,GstRTSPSrc))
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#define GST_RTSPSRC_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSPSRC,GstRTSPSrcClass))
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#define GST_IS_RTSPSRC(obj) \
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(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSPSRC))
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#define GST_IS_RTSPSRC_CLASS(klass) \
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(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSPSRC))
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#define GST_RTSPSRC_CAST(obj) \
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((GstRTSPSrc *)(obj))
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typedef struct _GstRTSPSrc GstRTSPSrc;
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typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
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#define GST_RTSP_STATE_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->state_rec_lock)
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#define GST_RTSP_STATE_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
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#define GST_RTSP_STATE_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
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#define GST_RTSP_STREAM_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->stream_rec_lock)
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#define GST_RTSP_STREAM_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
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#define GST_RTSP_STREAM_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
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#define GST_RTSP_CONN_GET_LOCK(rtsp) (GST_RTSPSRC_CAST(rtsp)->conn_rec_lock)
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#define GST_RTSP_CONN_LOCK(rtsp) (g_static_rec_mutex_lock (GST_RTSP_CONN_GET_LOCK(rtsp)))
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#define GST_RTSP_CONN_UNLOCK(rtsp) (g_static_rec_mutex_unlock (GST_RTSP_CONN_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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};
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typedef struct _GstRTSPStream GstRTSPStream;
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struct _GstRTSPStream {
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gint id;
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GstRTSPSrc *parent; /* parent, no extra ref to parent is taken */
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/* pad we expose or NULL when it does not have an actual pad */
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GstPad *srcpad;
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GstFlowReturn last_ret;
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gboolean added;
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gboolean disabled;
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gboolean eos;
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gboolean discont;
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/* for interleaved mode */
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guint8 channel[2];
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GstCaps *caps;
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GstPad *channelpad[2];
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/* our udp sources */
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GstElement *udpsrc[2];
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GstPad *blockedpad;
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gboolean is_ipv6;
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/* our udp sinks back to the server */
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GstElement *udpsink[2];
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GstPad *rtcppad;
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/* fakesrc for sending dummy data */
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GstElement *fakesrc;
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/* state */
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gint pt;
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guint port;
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gboolean container;
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/* original control url */
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gchar *control_url;
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guint32 ssrc;
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guint32 seqbase;
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guint64 timebase;
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/* per stream connection */
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GstRTSPConnInfo conninfo;
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/* bandwidth */
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guint as_bandwidth;
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guint rs_bandwidth;
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guint rr_bandwidth;
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/* destination */
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gchar *destination;
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guint ttl;
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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 _GstRTSPSrc {
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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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GStaticRecMutex *stream_rec_lock;
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GstSegment segment;
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gboolean running;
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gboolean need_range;
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gboolean skip;
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gint free_channel;
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GstEvent *close_segment;
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GstEvent *start_segment;
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GstClockTime base_time;
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/* UDP mode loop */
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gint loop_cmd;
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gboolean ignore_timeout;
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/* mutex for protecting state changes */
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GStaticRecMutex *state_rec_lock;
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/* mutex for protecting the connection */
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GStaticRecMutex *conn_rec_lock;
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GstSDPMessage *sdp;
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gboolean from_sdp;
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gint numstreams;
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GList *streams;
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GstStructure *props;
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gboolean need_activate;
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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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GTimeVal tcp_timeout;
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GTimeVal *ptcp_timeout;
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guint latency;
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guint connection_speed;
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GstRTSPNatMethod nat_method;
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gboolean do_rtcp;
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gchar *proxy_host;
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guint proxy_port;
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gchar *proxy_user;
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gchar *proxy_passwd;
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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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gint buffer_mode;
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GstRTSPRange client_port_range;
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gint udp_buffer_size;
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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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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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/* supported methods */
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gint methods;
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/* session management */
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GstElement *session;
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gulong session_sig_id;
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gulong session_ptmap_id;
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GstRTSPConnInfo conninfo;
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/* a list of RTSP extensions as GstElement */
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GstRTSPExtensionList *extensions;
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};
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struct _GstRTSPSrcClass {
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GstBinClass parent_class;
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};
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GType gst_rtspsrc_get_type(void);
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G_END_DECLS
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#endif /* __GST_RTSPSRC_H__ */
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