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gstreamer/gst/rtsp-server/rtsp-stream.h

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rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.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
Fix FSF address
2012-11-04 00:14:25 +00:00
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
*/
#include <gst/gst.h>
rtsp-server: Use single-include rtsp header to make sure we get all definitions
2015-05-20 14:05:47 +00:00
#include <gst/rtsp/rtsp.h>
rtspstream: handle both ipv4 and ipv6 clients Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2013-05-30 06:07:48 +00:00
#include <gio/gio.h>
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
#ifndef __GST_RTSP_STREAM_H__
#define __GST_RTSP_STREAM_H__
G_BEGIN_DECLS
/* types for the media stream */
#define GST_TYPE_RTSP_STREAM (gst_rtsp_stream_get_type ())
#define GST_IS_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM))
#define GST_IS_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM))
#define GST_RTSP_STREAM_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
#define GST_RTSP_STREAM(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStream))
#define GST_RTSP_STREAM_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM, GstRTSPStreamClass))
#define GST_RTSP_STREAM_CAST(obj) ((GstRTSPStream*)(obj))
#define GST_RTSP_STREAM_CLASS_CAST(klass) ((GstRTSPStreamClass*)(klass))
typedef struct _GstRTSPStream GstRTSPStream;
typedef struct _GstRTSPStreamClass GstRTSPStreamClass;
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
typedef struct _GstRTSPStreamPrivate GstRTSPStreamPrivate;
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
#include "rtsp-stream-transport.h"
stream: add methods to deal with address pool Add methods to get and set the address pool for the stream Add method to allocate and get the multicast addresses for this stream.
2012-11-15 14:36:21 +00:00
#include "rtsp-address-pool.h"
stream: add method to filter transports Add a method to safely iterate and collect the stream transports Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2013-11-18 10:18:15 +00:00
#include "rtsp-session.h"
rtsp-server: Implement clock signalling according to RFC7273 For NTP and PTP clocks we signal the actual clock that is used and signal the direct media clock offset. For all other clocks we at least signal that it's the local sender clock. This allows receivers to know which clock was used to generate the media and its RTP timestamps. Receivers can then implement network synchronization, either absolute or at least relative by getting the sender clock rate directly via NTP/PTP instead of estimating it from RTP timestamps and packet receive times. https://bugzilla.gnome.org/show_bug.cgi?id=760005
2015-12-30 16:39:05 +00:00
#include "rtsp-media.h"
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
/**
* GstRTSPStream:
*
rtsp: make object details private Make all object details private Add methods to access private bits
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* The definition of a media stream.
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
*/
struct _GstRTSPStream {
GObject parent;
docs: more updates
2013-07-11 10:24:33 +00:00
/*< private >*/
rtsp: make object details private Make all object details private Add methods to access private bits
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GstRTSPStreamPrivate *priv;
rtsp-stream: minor code formatting fix
2015-02-11 17:24:38 +00:00
gpointer _gst_reserved[GST_PADDING];
};
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
struct _GstRTSPStreamClass {
GObjectClass parent_class;
rtsp-server: add padding to many public structures Not mini objects though, since they are not subclassable anyway, nor kept on the stack or inlined in a structure.
2013-12-12 00:36:07 +00:00
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
};
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GType gst_rtsp_stream_get_type (void);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GstRTSPStream * gst_rtsp_stream_new (guint idx, GstElement *payloader,
Add initial support for RECORD We currently only support media that is RECORD or PLAY only, not both at once. https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-09 11:40:47 +00:00
GstPad *pad);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
guint gst_rtsp_stream_get_index (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp-stream: add getter for payload type * gst/rtsp-server/rtsp-stream.c: add new method gst_rtsp_stream_get_pt. * gst/rtsp-server/rtsp-media.c (pad_added_cb): find real payloader element and create the stream with this one instead of the dynpay%d element. https://bugzilla.gnome.org/show_bug.cgi?id=712396
2013-11-15 20:14:32 +00:00
guint gst_rtsp_stream_get_pt (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add method to get the srcpad
2013-04-22 15:32:31 +00:00
GstPad * gst_rtsp_stream_get_srcpad (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
Add initial support for RECORD We currently only support media that is RECORD or PLAY only, not both at once. https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-09 11:40:47 +00:00
GstPad * gst_rtsp_stream_get_sinkpad (GstRTSPStream *stream);
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: add methods and property to set control string
2013-07-02 12:44:35 +00:00
void gst_rtsp_stream_set_control (GstRTSPStream *stream, const gchar *control);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add methods and property to set control string
2013-07-02 12:44:35 +00:00
gchar * gst_rtsp_stream_get_control (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add method to check control url of stream
2013-07-03 13:13:45 +00:00
gboolean gst_rtsp_stream_has_control (GstRTSPStream *stream, const gchar *control);
stream: add methods and property to set control string
2013-07-02 12:44:35 +00:00
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
void gst_rtsp_stream_set_mtu (GstRTSPStream *stream, guint mtu);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
guint gst_rtsp_stream_get_mtu (GstRTSPStream *stream);
rtsp: fix MTU setting Fix setting of the MTU. There is no need for a vmethod.
2012-10-26 10:04:02 +00:00
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
dscp qos support in gst-rtsp-stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2013-06-24 08:43:59 +00:00
void gst_rtsp_stream_set_dscp_qos (GstRTSPStream *stream, gint dscp_qos);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
dscp qos support in gst-rtsp-stream Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702645
2013-06-24 08:43:59 +00:00
gint gst_rtsp_stream_get_dscp_qos (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add method to check supported transport Add a method to check if a transport is supported
2014-01-07 11:14:15 +00:00
gboolean gst_rtsp_stream_is_transport_supported (GstRTSPStream *stream,
GstRTSPTransport *transport);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add property to configure profiles
2014-01-07 11:21:09 +00:00
void gst_rtsp_stream_set_profiles (GstRTSPStream *stream, GstRTSPProfile profiles);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add property to configure profiles
2014-01-07 11:21:09 +00:00
GstRTSPProfile gst_rtsp_stream_get_profiles (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add protocols property
2013-08-16 14:08:43 +00:00
void gst_rtsp_stream_set_protocols (GstRTSPStream *stream, GstRTSPLowerTrans protocols);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add protocols property
2013-08-16 14:08:43 +00:00
GstRTSPLowerTrans gst_rtsp_stream_get_protocols (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add methods to deal with address pool Add methods to get and set the address pool for the stream Add method to allocate and get the multicast addresses for this stream.
2012-11-15 14:36:21 +00:00
void gst_rtsp_stream_set_address_pool (GstRTSPStream *stream, GstRTSPAddressPool *pool);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add methods to deal with address pool Add methods to get and set the address pool for the stream Add method to allocate and get the multicast addresses for this stream.
2012-11-15 14:36:21 +00:00
GstRTSPAddressPool *
gst_rtsp_stream_get_address_pool (GstRTSPStream *stream);
rtsp-media: Add support for setting the multicast interface https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-02 17:42:13 +00:00
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp-media: Add support for setting the multicast interface https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-02 17:42:13 +00:00
void gst_rtsp_stream_set_multicast_iface (GstRTSPStream *stream, const gchar * multicast_iface);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp-media: Add support for setting the multicast interface https://bugzilla.gnome.org/show_bug.cgi?id=763000
2016-03-02 17:42:13 +00:00
gchar * gst_rtsp_stream_get_multicast_iface (GstRTSPStream *stream);
stream: add methods to deal with address pool Add methods to get and set the address pool for the stream Add method to allocate and get the multicast addresses for this stream.
2012-11-15 14:36:21 +00:00
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
client: Check client provided addresses against the address pool
2013-02-13 23:32:20 +00:00
GstRTSPAddress * gst_rtsp_stream_reserve_address (GstRTSPStream *stream,
const gchar * address,
guint port,
guint n_ports,
guint ttl);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
gboolean gst_rtsp_stream_join_bin (GstRTSPStream *stream,
stream: improve join and leave of the pipeline simplify code Do the cleanup properly Add some docs
2012-10-26 15:28:10 +00:00
GstBin *bin, GstElement *rtpbin,
GstState state);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
gboolean gst_rtsp_stream_leave_bin (GstRTSPStream *stream,
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GstBin *bin, GstElement *rtpbin);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp-server: Implement clock signalling according to RFC7273 For NTP and PTP clocks we signal the actual clock that is used and signal the direct media clock offset. For all other clocks we at least signal that it's the local sender clock. This allows receivers to know which clock was used to generate the media and its RTP timestamps. Receivers can then implement network synchronization, either absolute or at least relative by getting the sender clock rate directly via NTP/PTP instead of estimating it from RTP timestamps and packet receive times. https://bugzilla.gnome.org/show_bug.cgi?id=760005
2015-12-30 16:39:05 +00:00
GstBin * gst_rtsp_stream_get_joined_bin (GstRTSPStream *stream);
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: add API to block streams Add an API to block on the streams and make it post a message. Based on patch by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 12:42:21 +00:00
gboolean gst_rtsp_stream_set_blocked (GstRTSPStream * stream,
gboolean blocked);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
stream: add API to block streams Add an API to block on the streams and make it post a message. Based on patch by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 12:42:21 +00:00
gboolean gst_rtsp_stream_is_blocking (GstRTSPStream * stream);
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
GST_EXPORT
gboolean gst_rtsp_stream_unblock_linked (GstRTSPStream * stream);
Mark symbols explicitly for export with GST_EXPORT
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GST_EXPORT
rtsp-stream: Add functions for using rtsp-stream from the client Add a boolean to indicate that the rtsp-stream is running on the 'client' side of an RTSP connection, for sending streams via RECORD. In that case, the roles of the client/server ports in transport setup are swapped. https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-16 14:12:28 +00:00
void gst_rtsp_stream_set_client_side (GstRTSPStream *stream, gboolean client_side);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-stream: Add functions for using rtsp-stream from the client Add a boolean to indicate that the rtsp-stream is running on the 'client' side of an RTSP connection, for sending streams via RECORD. In that case, the roles of the client/server ports in transport setup are swapped. https://bugzilla.gnome.org/show_bug.cgi?id=758180
2015-11-16 14:12:28 +00:00
gboolean gst_rtsp_stream_is_client_side (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
void gst_rtsp_stream_get_server_port (GstRTSPStream *stream,
rtspstream: handle both ipv4 and ipv6 clients Fixes https://bugzilla.gnome.org/show_bug.cgi?id=701129
2013-05-30 06:07:48 +00:00
GstRTSPRange *server_port,
GSocketFamily family);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: add more support for IPv6 Rename _get_address to _get_multicast_address in GstRTSPStream to make it clear that this function only deals with multicast. Make it possible to have both an IPv4 and IPv6 multicast address on a stream. Give the client an IPv4 or IPv6 address depending on the address it used to connect to the server. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702002
2013-07-01 14:46:07 +00:00
GstRTSPAddress * gst_rtsp_stream_get_multicast_address (GstRTSPStream *stream,
GSocketFamily family);
stream: allow access to the rtp session https://bugzilla.gnome.org/show_bug.cgi?id=703004
2013-06-24 18:41:27 +00:00
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: allow access to the rtp session https://bugzilla.gnome.org/show_bug.cgi?id=703004
2013-06-24 18:41:27 +00:00
GObject * gst_rtsp_stream_get_rtpsession (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
sdp: add rollover counters for all sender SSRC We add different crypto sessions in MIKEY, one for each sender SSRC. Currently, all of them will have the same security policy, 0. The rollover counters are obtained from the srtpenc element using the "stats" property. https://bugzilla.gnome.org/show_bug.cgi?id=730539
2016-04-15 05:56:11 +00:00
GstElement * gst_rtsp_stream_get_srtp_encoder (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
void gst_rtsp_stream_get_ssrc (GstRTSPStream *stream,
guint *ssrc);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
gboolean gst_rtsp_stream_get_rtpinfo (GstRTSPStream *stream,
stream: also return the running-time Return the running-time in the rtpinfo as well.
2013-12-26 13:43:35 +00:00
guint *rtptime, guint *seq,
stream: return clock-rate from get_rtpinfo And use it to correct the rtptime to the requested start-time. See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 16:02:50 +00:00
guint *clock_rate,
stream: also return the running-time Return the running-time in the rtpinfo as well.
2013-12-26 13:43:35 +00:00
GstClockTime *running_time);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
GstCaps * gst_rtsp_stream_get_caps (GstRTSPStream *stream);
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GstFlowReturn gst_rtsp_stream_recv_rtp (GstRTSPStream *stream,
GstBuffer *buffer);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
GstFlowReturn gst_rtsp_stream_recv_rtcp (GstRTSPStream *stream,
GstBuffer *buffer);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
gboolean gst_rtsp_stream_add_transport (GstRTSPStream *stream,
GstRTSPStreamTransport *trans);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
gboolean gst_rtsp_stream_remove_transport (GstRTSPStream *stream,
GstRTSPStreamTransport *trans);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
GSocket * gst_rtsp_stream_get_rtp_socket (GstRTSPStream *stream,
GSocketFamily family);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: Add functions to get rtp and rtcp sockets Fixes https://bugzilla.gnome.org/show_bug.cgi?id=710100
2013-11-12 11:04:55 +00:00
GSocket * gst_rtsp_stream_get_rtcp_socket (GstRTSPStream *stream,
GSocketFamily family);
rtsp-stream: add functions to get rtp and rtcp multicast sockets Change-Id: Iddfe6e0bd250cb0159096d5eba9e4202d22b56db https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-16 09:15:55 +00:00
GST_EXPORT
GSocket * gst_rtsp_stream_get_rtp_multicast_socket (GstRTSPStream *stream,
GSocketFamily family);
GST_EXPORT
GSocket * gst_rtsp_stream_get_rtcp_multicast_socket (GstRTSPStream *stream,
GSocketFamily family);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: add method to set crypto info Make a method to configure the crypto information of a stream. Set udpsrc in READY instead of PAUSED so that we can configure caps later.
2014-04-02 10:36:16 +00:00
gboolean gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
guint ssrc, GstCaps * crypto);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-media: Query position and stop time only on the RTP parts of the pipeline The RTCP parts, in specific the RTCP udpsinks, are not flushed when seeking and will always continue counting the time. This leads to the NPT after a backwards seek to be something completely different to the actual seek position. https://bugzilla.gnome.org/show_bug.cgi?id=732644
2014-08-05 14:12:19 +00:00
gboolean gst_rtsp_stream_query_position (GstRTSPStream * stream,
gint64 * position);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-media: Query position and stop time only on the RTP parts of the pipeline The RTCP parts, in specific the RTCP udpsinks, are not flushed when seeking and will always continue counting the time. This leads to the NPT after a backwards seek to be something completely different to the actual seek position. https://bugzilla.gnome.org/show_bug.cgi?id=732644
2014-08-05 14:12:19 +00:00
gboolean gst_rtsp_stream_query_stop (GstRTSPStream * stream,
gint64 * stop);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-media: Make sure that sequence numbers are monotonic after pause The sequence number is not monotonic for RTP packets after pause. The reason is basepayloader generates a randon sequence number when the pipeline goes from ready to pause. With this fix generation of sequence number will be monotonic when going from pause to play request. https://bugzilla.gnome.org/show_bug.cgi?id=736017
2014-09-08 07:26:23 +00:00
void gst_rtsp_stream_set_seqnum_offset (GstRTSPStream *stream, guint16 seqnum);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-media: Make sure that sequence numbers are monotonic after pause The sequence number is not monotonic for RTP packets after pause. The reason is basepayloader generates a randon sequence number when the pipeline goes from ready to pause. With this fix generation of sequence number will be monotonic when going from pause to play request. https://bugzilla.gnome.org/show_bug.cgi?id=736017
2014-09-08 07:26:23 +00:00
guint16 gst_rtsp_stream_get_current_seqnum (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
media: implement ssrc-multiplexed retransmission support based off RFC 4588 and the server-rtpaux example in -good
2014-11-26 14:12:36 +00:00
void gst_rtsp_stream_set_retransmission_time (GstRTSPStream *stream, GstClockTime time);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
media: implement ssrc-multiplexed retransmission support based off RFC 4588 and the server-rtpaux example in -good
2014-11-26 14:12:36 +00:00
GstClockTime gst_rtsp_stream_get_retransmission_time (GstRTSPStream *stream);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
media: implement ssrc-multiplexed retransmission support based off RFC 4588 and the server-rtpaux example in -good
2014-11-26 14:12:36 +00:00
guint gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
media: implement ssrc-multiplexed retransmission support based off RFC 4588 and the server-rtpaux example in -good
2014-11-26 14:12:36 +00:00
void gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream,
guint rtx_pt);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-stream: Implement UDP buffer size setting. Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the UDP TX buffer size. Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com> Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-08-31 12:48:34 +00:00
void gst_rtsp_stream_set_buffer_size (GstRTSPStream *stream, guint size);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-stream: Implement UDP buffer size setting. Add gst_rtsp_stream_(get|set)_buffer_size and use it to configure the UDP TX buffer size. Incorporates a patch by Hyunjun Ko <zzoon.ko@samsung.com> Fixes https://bugzilla.gnome.org/show_bug.cgi?id=749095
2015-08-31 12:48:34 +00:00
guint gst_rtsp_stream_get_buffer_size (GstRTSPStream *stream);
rtsp-media: Make sure that sequence numbers are monotonic after pause The sequence number is not monotonic for RTP packets after pause. The reason is basepayloader generates a randon sequence number when the pipeline goes from ready to pause. With this fix generation of sequence number will be monotonic when going from pause to play request. https://bugzilla.gnome.org/show_bug.cgi?id=736017
2014-09-08 07:26:23 +00:00
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
Add initial support for RECORD We currently only support media that is RECORD or PLAY only, not both at once. https://bugzilla.gnome.org/show_bug.cgi?id=743175
2015-01-09 11:40:47 +00:00
void gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-stream: fix to get valid each stream data for request-aux-sender signal Because of duplicated g_signal_connect for request-aux-sender signal, wrong stream pointer is passed to the signal handler. Instead of passing each stream, pass stream array and get the relevant stream. https://bugzilla.gnome.org/show_bug.cgi?id=747839
2015-04-15 00:45:23 +00:00
GstElement * gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid);
rtsp-stream: postpone UDP socket allocation until SETUP Postpone the allocation of the UDP sockets until we know what transport has been chosen by the client. Both unicast and multicast UDP sources are created in one function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 13:59:32 +00:00
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-stream: postpone UDP socket allocation until SETUP Postpone the allocation of the UDP sockets until we know what transport has been chosen by the client. Both unicast and multicast UDP sources are created in one function. https://bugzilla.gnome.org/show_bug.cgi?id=757488
2016-02-23 13:59:32 +00:00
gboolean gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream, GSocketFamily family,
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
GstRTSPTransport *transport, gboolean use_client_settings);
rtsp-server: Implement clock signalling according to RFC7273 For NTP and PTP clocks we signal the actual clock that is used and signal the direct media clock offset. For all other clocks we at least signal that it's the local sender clock. This allows receivers to know which clock was used to generate the media and its RTP timestamps. Receivers can then implement network synchronization, either absolute or at least relative by getting the sender clock rate directly via NTP/PTP instead of estimating it from RTP timestamps and packet receive times. https://bugzilla.gnome.org/show_bug.cgi?id=760005
2015-12-30 16:39:05 +00:00
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-server: Implement clock signalling according to RFC7273 For NTP and PTP clocks we signal the actual clock that is used and signal the direct media clock offset. For all other clocks we at least signal that it's the local sender clock. This allows receivers to know which clock was used to generate the media and its RTP timestamps. Receivers can then implement network synchronization, either absolute or at least relative by getting the sender clock rate directly via NTP/PTP instead of estimating it from RTP timestamps and packet receive times. https://bugzilla.gnome.org/show_bug.cgi?id=760005
2015-12-30 16:39:05 +00:00
void gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream, GstRTSPPublishClockMode mode);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
rtsp-server: Implement clock signalling according to RFC7273 For NTP and PTP clocks we signal the actual clock that is used and signal the direct media clock offset. For all other clocks we at least signal that it's the local sender clock. This allows receivers to know which clock was used to generate the media and its RTP timestamps. Receivers can then implement network synchronization, either absolute or at least relative by getting the sender clock rate directly via NTP/PTP instead of estimating it from RTP timestamps and packet receive times. https://bugzilla.gnome.org/show_bug.cgi?id=760005
2015-12-30 16:39:05 +00:00
GstRTSPPublishClockMode gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream);
Dynamically reconfigure pipeline in PLAY based on transports The initial pipeline does not contain specific transport elements. The receiver and the sender parts are added after PLAY. If the media is shared, the streams are dynamically reconfigured after each PLAY. https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-17 08:44:33 +00:00
GST_EXPORT
gboolean gst_rtsp_stream_complete_stream (GstRTSPStream * stream, const GstRTSPTransport * transport);
rtsp-media: seek on media pipelines that are complete Make sure that a seek is performed on pipelines that contain at least one sink element. Change-Id: Icf398e10add3191d104b1289de612412da326819 https://bugzilla.gnome.org/show_bug.cgi?id=788340
2017-10-23 07:49:09 +00:00
GST_EXPORT
gboolean gst_rtsp_stream_is_complete (GstRTSPStream * stream);
stream: add method to filter transports Add a method to safely iterate and collect the stream transports Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2013-11-18 10:18:15 +00:00
/**
* GstRTSPStreamTransportFilterFunc:
* @stream: a #GstRTSPStream object
* @trans: a #GstRTSPStreamTransport in @stream
* @user_data: user data that has been given to gst_rtsp_stream_transport_filter()
*
* This function will be called by the gst_rtsp_stream_transport_filter(). An
* implementation should return a value of #GstRTSPFilterResult.
*
* When this function returns #GST_RTSP_FILTER_REMOVE, @trans will be removed
* from @stream.
*
* A return value of #GST_RTSP_FILTER_KEEP will leave @trans untouched in
* @stream.
*
* A value of #GST_RTSP_FILTER_REF will add @trans to the result #GList of
* gst_rtsp_stream_transport_filter().
*
* Returns: a #GstRTSPFilterResult.
*/
typedef GstRTSPFilterResult (*GstRTSPStreamTransportFilterFunc) (GstRTSPStream *stream,
GstRTSPStreamTransport *trans,
gpointer user_data);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
GST_EXPORT
stream: add method to filter transports Add a method to safely iterate and collect the stream transports Fixes https://bugzilla.gnome.org/show_bug.cgi?id=711664
2013-11-18 10:18:15 +00:00
GList * gst_rtsp_stream_transport_filter (GstRTSPStream *stream,
GstRTSPStreamTransportFilterFunc func,
gpointer user_data);
rtsp-server: Add g_autoptr() support to all types https://bugzilla.gnome.org/show_bug.cgi?id=754464
2015-11-10 19:17:18 +00:00
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPStream, gst_object_unref)
#endif
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
G_END_DECLS
#endif /* __GST_RTSP_STREAM_H__ */
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