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gstreamer/subprojects/gst-rtsp-server/gst/rtsp-server/rtsp-media.h

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Initial import
2008-10-09 12:29:12 +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.
Initial import
2008-10-09 12:29:12 +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>
media: add GstNetTimeProvider support Add a property to let the media provide a GstNetTimeProvider for its clock. Make methods to get the clock and nettimeprovider Add a x-gst-clock property to the SDP with the IP and port number of the nettime provider and also the current time of the clock. This should make it possible for (GStreamer) clients to slave their clock to the server clock.
2013-04-09 20:35:28 +00:00
#include <gst/net/gstnet.h>
Initial import
2008-10-09 12:29:12 +00:00
#ifndef __GST_RTSP_MEDIA_H__
#define __GST_RTSP_MEDIA_H__
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
#include "rtsp-server-prelude.h"
Initial import
2008-10-09 12:29:12 +00:00
G_BEGIN_DECLS
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
/* types for the media */
#define GST_TYPE_RTSP_MEDIA (gst_rtsp_media_get_type ())
#define GST_IS_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_MEDIA))
#define GST_IS_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_MEDIA))
#define GST_RTSP_MEDIA_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMedia))
#define GST_RTSP_MEDIA_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_MEDIA, GstRTSPMediaClass))
#define GST_RTSP_MEDIA_CAST(obj) ((GstRTSPMedia*)(obj))
#define GST_RTSP_MEDIA_CLASS_CAST(klass) ((GstRTSPMediaClass*)(klass))
Initial import
2008-10-09 12:29:12 +00:00
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
typedef struct _GstRTSPMedia GstRTSPMedia;
typedef struct _GstRTSPMediaClass GstRTSPMediaClass;
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 _GstRTSPMediaPrivate GstRTSPMediaPrivate;
Initial import
2008-10-09 12:29:12 +00:00
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
/**
* GstRTSPMediaStatus:
* @GST_RTSP_MEDIA_STATUS_UNPREPARED: media pipeline not prerolled
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
* @GST_RTSP_MEDIA_STATUS_UNPREPARING: media pipeline is busy doing a clean
* shutdown.
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
* @GST_RTSP_MEDIA_STATUS_PREPARING: media pipeline is prerolling
* @GST_RTSP_MEDIA_STATUS_PREPARED: media pipeline is prerolled
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
* @GST_RTSP_MEDIA_STATUS_SUSPENDED: media is suspended
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
* @GST_RTSP_MEDIA_STATUS_ERROR: media pipeline is in error
*
* The state of the media pipeline.
*/
typedef enum {
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
GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
GST_RTSP_MEDIA_STATUS_PREPARING = 2,
GST_RTSP_MEDIA_STATUS_PREPARED = 3,
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
GST_RTSP_MEDIA_STATUS_ERROR = 5
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
} GstRTSPMediaStatus;
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
/**
* GstRTSPSuspendMode:
* @GST_RTSP_SUSPEND_MODE_NONE: Media is not suspended
* @GST_RTSP_SUSPEND_MODE_PAUSE: Media is PAUSED in suspend
* @GST_RTSP_SUSPEND_MODE_RESET: The media is set to NULL when suspended
*
* The suspend mode of the media pipeline. A media pipeline is suspended right
media: correct misspelled words in description https://bugzilla.gnome.org/show_bug.cgi?id=733244
2014-07-16 11:39:42 +00:00
* after creating the SDP and when the client performs a PAUSED request.
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
*/
typedef enum {
GST_RTSP_SUSPEND_MODE_NONE = 0,
GST_RTSP_SUSPEND_MODE_PAUSE = 1,
GST_RTSP_SUSPEND_MODE_RESET = 2
} GstRTSPSuspendMode;
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:54:01 +00:00
/**
* GstRTSPTransportMode:
* @GST_RTSP_TRANSPORT_MODE_PLAY: Transport supports PLAY mode
* @GST_RTSP_TRANSPORT_MODE_RECORD: Transport supports RECORD mode
*
* The supported modes of the media.
*/
typedef enum {
GST_RTSP_TRANSPORT_MODE_PLAY = 1,
GST_RTSP_TRANSPORT_MODE_RECORD = 2,
} GstRTSPTransportMode;
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_PUBLISH_CLOCK_MODE_NONE: Publish nothing
* @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK: Publish the clock but not the offset
* @GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET: Publish the clock and offset
*
* Whether the clock and possibly RTP/clock offset should be published according to RFC7273.
*/
typedef enum {
GST_RTSP_PUBLISH_CLOCK_MODE_NONE,
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK,
GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET
} GstRTSPPublishClockMode;
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:54:01 +00:00
#define GST_TYPE_RTSP_TRANSPORT_MODE (gst_rtsp_transport_mode_get_type())
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:54:01 +00:00
GType gst_rtsp_transport_mode_get_type (void);
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
GType gst_rtsp_suspend_mode_get_type (void);
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
#define GST_TYPE_RTSP_PUBLISH_CLOCK_MODE (gst_rtsp_publish_clock_mode_get_type())
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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
GType gst_rtsp_publish_clock_mode_get_type (void);
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
#include "rtsp-stream.h"
#include "rtsp-thread-pool.h"
#include "rtsp-permissions.h"
#include "rtsp-address-pool.h"
media: add setup_sdp vmethod gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public gst_rtsp_media_setup_sdp. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2013-12-18 15:37:27 +00:00
#include "rtsp-sdp.h"
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
Add comments. Remove unused field
2009-01-20 18:43:47 +00:00
/**
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
* GstRTSPMedia:
Add comments. Remove unused field
2009-01-20 18:43:47 +00:00
*
Cleanups and doc updates Add some more documentation and do some minor cleanups here and there.
2009-01-22 17:35:17 +00:00
* A class that contains the GStreamer element along with a list of
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 objects that can produce data.
Cleanups and doc updates Add some more documentation and do some minor cleanups here and there.
2009-01-22 17:35:17 +00:00
*
* This object is usually created from a #GstRTSPMediaFactory.
Add comments. Remove unused field
2009-01-20 18:43:47 +00:00
*/
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
struct _GstRTSPMedia {
media: avoid doing _get_state() for state changes When preparing, use the ASYNC_DONE and ERROR messages in the bus handler to wait until the media is prerolled or in error. This avoids doing a blocking call of gst_element_get_state() that can cause lockups when there is an error. Fixes #611899
2010-03-05 16:51:26 +00:00
GObject parent;
Initial import
2008-10-09 12:29:12 +00:00
docs: update docs
2013-07-11 10:18:26 +00:00
/*< private >*/
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
GstRTSPMediaPrivate *priv;
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
gpointer _gst_reserved[GST_PADDING];
Initial import
2008-10-09 12:29:12 +00:00
};
Handle media bus messages Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 15:39:36 +00:00
/**
* GstRTSPMediaClass:
* @handle_message: handle a message
media: make media_prepare virtual Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2014-04-09 14:44:21 +00:00
* @prepare: the default implementation adds all elements and sets the
* pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING
* in case of NO_PREROLL elements).
Added vmethod unprepare to GstRTSPMedia The default implementation sets the state of the pipeline to GST_STATE_NULL
2009-06-12 16:05:30 +00:00
* @unprepare: the default implementation sets the pipeline's state
docs: update docs
2012-10-26 10:33:21 +00:00
* to GST_STATE_NULL and removes all elements.
media: Make suspend()/unsuspend() virtual Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2014-05-14 07:31:31 +00:00
* @suspend: the default implementation sets the pipeline's state to
* GST_STATE_NULL GST_STATE_PAUSED depending on the selected
* suspend mode.
* @unsuspend: the default implementation reverts the suspend operation.
* The pipeline will be prerolled again if it's state was
* set to GST_STATE_NULL in suspend.
docs: more updates
2013-07-11 10:24:33 +00:00
* @convert_range: convert a range to the given unit
media: correct misspelled words in description https://bugzilla.gnome.org/show_bug.cgi?id=733244
2014-07-16 11:39:42 +00:00
* @query_position: query the current position in the pipeline
docs: more updates
2013-07-11 10:24:33 +00:00
* @query_stop: query when playback will stop
Handle media bus messages Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 15:39:36 +00:00
*
* The RTSP media class
*/
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
struct _GstRTSPMediaClass {
Initial import
2008-10-09 12:29:12 +00:00
GObjectClass parent_class;
Handle media bus messages Handle media bus messages in a custom mainloop and dispatch them to the RTSPMedia objects. Let the default implementation handle some common messages.
2009-02-13 15:39:36 +00:00
media: add signal to notify of unprepare
2009-04-03 20:45:57 +00:00
/* vmethods */
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 (*handle_message) (GstRTSPMedia *media, GstMessage *message);
media: make media_prepare virtual Fixes https://bugzilla.gnome.org/show_bug.cgi?id=728029
2014-04-09 14:44:21 +00:00
gboolean (*prepare) (GstRTSPMedia *media, GstRTSPThread *thread);
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 (*unprepare) (GstRTSPMedia *media);
media: Make suspend()/unsuspend() virtual Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730109
2014-05-14 07:31:31 +00:00
gboolean (*suspend) (GstRTSPMedia *media);
gboolean (*unsuspend) (GstRTSPMedia *media);
media: convert_range replaces get_range_times get_range_times worked for handling UTC ranges for seeks, but we also need to convert back from NPT to the requested unit in get_range_string. convert_range is now used for both. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702084
2013-06-05 13:49:45 +00:00
gboolean (*convert_range) (GstRTSPMedia *media, GstRTSPTimeRange *range,
GstRTSPRangeUnit unit);
Add query_position and query_stop vmethods to rtsp-media
2013-06-10 21:32:40 +00:00
gboolean (*query_position) (GstRTSPMedia *media, gint64 *position);
gboolean (*query_stop) (GstRTSPMedia *media, gint64 *stop);
media: add new create_rtpbin vmethod * gst/rtsp-server/rtsp-media.[ch]: add new create_rtpbin vmethod. https://bugzilla.gnome.org/show_bug.cgi?id=719734
2013-12-03 19:54:42 +00:00
GstElement * (*create_rtpbin) (GstRTSPMedia *media);
Add vmethod for rtsp-media subclass to access rtpbin
2013-06-11 23:10:01 +00:00
gboolean (*setup_rtpbin) (GstRTSPMedia *media, GstElement *rtpbin);
media: add setup_sdp vmethod gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public gst_rtsp_media_setup_sdp. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2013-12-18 15:37:27 +00:00
gboolean (*setup_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp, GstSDPInfo *info);
media: add signal to notify of unprepare
2009-04-03 20:45:57 +00:00
/* signals */
media: listen to pad-removed signals Listen to the pad-removed signal and remove the stream associated with the removed pad. Add signal to be notified of the removed pad. Remove the fakesink in unprepare() Fix signatures of the signal methods
2013-04-22 15:34:37 +00:00
void (*new_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
void (*removed_stream) (GstRTSPMedia *media, GstRTSPStream * stream);
media: add signal for new streams This allows applications to listen for new streams and configure properties on them, like the address pool.
2012-11-15 14:41:42 +00:00
media: listen to pad-removed signals Listen to the pad-removed signal and remove the stream associated with the removed pad. Add signal to be notified of the removed pad. Remove the fakesink in unprepare() Fix signatures of the signal methods
2013-04-22 15:34:37 +00:00
void (*prepared) (GstRTSPMedia *media);
void (*unprepared) (GstRTSPMedia *media);
media: add signal to notify of state changes
2010-12-28 17:34:10 +00:00
media: add signal to notify of pending state changes
2014-01-21 11:01:25 +00:00
void (*target_state) (GstRTSPMedia *media, GstState state);
media: listen to pad-removed signals Listen to the pad-removed signal and remove the stream associated with the removed pad. Add signal to be notified of the removed pad. Remove the fakesink in unprepare() Fix signatures of the signal methods
2013-04-22 15:34:37 +00:00
void (*new_state) (GstRTSPMedia *media, GstState state);
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
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
gboolean (*handle_sdp) (GstRTSPMedia *media, GstSDPMessage *sdp);
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 >*/
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
gpointer _gst_reserved[GST_PADDING_LARGE-1];
Initial import
2008-10-09 12:29:12 +00:00
};
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
More improvements Rename GstRTSPMediaBin to GstRTSPMedia Parse the request url into a GstRTSPUri object and pass this object to the various handlers and methods that require the uri.
2009-01-22 16:58:19 +00:00
GType gst_rtsp_media_get_type (void);
Initial import
2008-10-09 12:29:12 +00:00
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
/* creating the media */
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
GstRTSPMedia * gst_rtsp_media_new (GstElement *element);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add _get_element() method Add method to get the element used when creating the media. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703008
2013-06-24 21:56:57 +00:00
GstElement * gst_rtsp_media_get_element (GstRTSPMedia *media);
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
GstRTSPMediaStatus gst_rtsp_media_get_status (GstRTSPMedia *media);
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: make it possible to set permissions Make it possible to set permissions on media and media factory objects
2013-07-09 12:33:43 +00:00
void gst_rtsp_media_set_permissions (GstRTSPMedia *media,
GstRTSPPermissions *permissions);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: make it possible to set permissions Make it possible to set permissions on media and media factory objects
2013-07-09 12:33:43 +00:00
GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
void gst_rtsp_media_set_shared (GstRTSPMedia *media, gboolean shared);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
Initial import
2008-10-09 12:29:12 +00:00
rtsp-server: media-factory: Make sure a shared media is actually still usable Previously it was possible that a shared media was just in the process of being unprepared because the last client disappeared, while another client retrieved it from the cache and then tried to use it. Unless the media was reusable this would've then failed unnecessarily. To avoid this it is necessary to lock the media directly in gst_rtsp_media_factory_construct() and return a locked media. After locking the cached media it is necessary to check if the media was ever unprepared or is actually reusable and based on that either reuse it or create a new media. This minimally changes the gst_rtsp_media_factory_construct() API to always return a locked media, and adds a new gst_rtsp_media_can_be_shared() function to check if a media can actually be shared in practice. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4606>
2023-05-11 18:09:25 +00:00
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_can_be_shared (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN Without TEARDOWN it might be desireable to keep the media running and continue sending data to the client, even if the RTSP connection itself is disconnected. Only do this for session medias that have only UDP transports. If there's at least on TCP transport, it will stop working and cause problems when the connection is disconnected. https://bugzilla.gnome.org/show_bug.cgi?id=758999
2015-12-09 16:24:24 +00:00
void gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia *media, gboolean stop_on_disconnect);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Add property to decide if sending media should be stopped when a client disconnects without TEARDOWN Without TEARDOWN it might be desireable to keep the media running and continue sending data to the client, even if the RTSP connection itself is disconnected. Only do this for session medias that have only UDP transports. If there's at least on TCP transport, it will stop working and cause problems when the connection is disconnected. https://bugzilla.gnome.org/show_bug.cgi?id=758999
2015-12-09 16:24:24 +00:00
gboolean gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:54:01 +00:00
void gst_rtsp_media_set_transport_mode (GstRTSPMedia *media, GstRTSPTransportMode mode);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Use flags to distinguish between PLAY and RECORD media
2015-01-28 17:54:01 +00:00
GstRTSPTransportMode gst_rtsp_media_get_transport_mode (GstRTSPMedia *media);
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
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
void gst_rtsp_media_set_reusable (GstRTSPMedia *media, gboolean reusable);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
media: allow configuration of allowed lower transport
2010-03-19 14:15:29 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
stream: add property to configure profiles
2014-01-07 11:21:09 +00:00
void gst_rtsp_media_set_profiles (GstRTSPMedia *media, GstRTSPProfile profiles);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
stream: add property to configure profiles
2014-01-07 11:21:09 +00:00
GstRTSPProfile gst_rtsp_media_get_profiles (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: allow configuration of allowed lower transport
2010-03-19 14:15:29 +00:00
void gst_rtsp_media_set_protocols (GstRTSPMedia *media, GstRTSPLowerTrans protocols);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: allow configuration of allowed lower transport
2010-03-19 14:15:29 +00:00
GstRTSPLowerTrans gst_rtsp_media_get_protocols (GstRTSPMedia *media);
media-factory: add eos-shutdown property Add an eos-shutdown property that will send an EOS to the pipeline before shutting it down. This allows for nice cleanup in case of a muxer. Fixes #625597
2010-08-20 16:17:08 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media-factory: add eos-shutdown property Add an eos-shutdown property that will send an EOS to the pipeline before shutting it down. This allows for nice cleanup in case of a muxer. Fixes #625597
2010-08-20 16:17:08 +00:00
void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media-factory: add eos-shutdown property Add an eos-shutdown property that will send an EOS to the pipeline before shutting it down. This allows for nice cleanup in case of a muxer. Fixes #625597
2010-08-20 16:17:08 +00:00
gboolean gst_rtsp_media_is_eos_shutdown (GstRTSPMedia *media);
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp: use AddressPool Remove the multicast_group property. Use the configured addresspool to allocate multicast addresses.
2012-11-14 16:23:59 +00:00
void gst_rtsp_media_set_address_pool (GstRTSPMedia *media, GstRTSPAddressPool *pool);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp: use AddressPool Remove the multicast_group property. Use the configured addresspool to allocate multicast addresses.
2012-11-14 16:23:59 +00:00
GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_set_multicast_iface (GstRTSPMedia *media, const gchar *multicast_iface);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_get_multicast_iface (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add property to configure kernel buffer sizes Add a property to configure the kernel UDP buffer size.
2011-01-31 16:28:22 +00:00
void gst_rtsp_media_set_buffer_size (GstRTSPMedia *media, guint size);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add property to configure kernel buffer sizes Add a property to configure the kernel UDP buffer size.
2011-01-31 16:28:22 +00:00
guint gst_rtsp_media_get_buffer_size (GstRTSPMedia *media);
rtsp-server: Add new ensure-keyunit-on-start property While the suspend modes NONE and PAUSED provided a low startup latency for connecting clients they did not ensure that streams started on fresh data. With this property we can maintain the low startup latency of those suspend modes while also ensuring that a stream starts on a key unit. Furthermore, by modifying the value of a new property, ensure-keyunit-on-start-timeout, it is possible to accept a keyunit of a certain age but discard it if too much time has passed and instead force a new keyunit. Fixes #2443 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4334>
2023-06-26 06:25:36 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_media_set_ensure_keyunit_on_start (GstRTSPMedia* media,
gboolean ensure_keyunit_on_start);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_ensure_keyunit_on_start (GstRTSPMedia* media);
GST_RTSP_SERVER_API
void gst_rtsp_media_set_ensure_keyunit_on_start_timeout (GstRTSPMedia* media,
guint timeout);
GST_RTSP_SERVER_API
guint gst_rtsp_media_get_ensure_keyunit_on_start_timeout (GstRTSPMedia* media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_set_retransmission_time (GstRTSPMedia *media, GstClockTime time);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_get_retransmission_time (GstRTSPMedia *media);
rtsp-server: add API to enable retransmission requests "do-retransmission" was previously set when rtx-time != 0, which made no sense as do-retransmission is used to enable the sending of retransmission requests, where as rtx-time is used by the peer to enable storing of buffers in order to respond to retransmission requests. rtsp-media now also provides a callback for the request-aux-receiver signal. https://bugzilla.gnome.org/show_bug.cgi?id=794822
2018-03-29 20:49:26 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
gboolean do_retransmission);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Expose latency setting for setting the rtpbin latency
2015-01-19 12:20:39 +00:00
void gst_rtsp_media_set_latency (GstRTSPMedia *media, guint latency);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Expose latency setting for setting the rtpbin latency
2015-01-19 12:20:39 +00:00
guint gst_rtsp_media_get_latency (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add GstNetTimeProvider support Add a property to let the media provide a GstNetTimeProvider for its clock. Make methods to get the clock and nettimeprovider Add a x-gst-clock property to the SDP with the IP and port number of the nettime provider and also the current time of the clock. This should make it possible for (GStreamer) clients to slave their clock to the server clock.
2013-04-09 20:35:28 +00:00
void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add GstNetTimeProvider support Add a property to let the media provide a GstNetTimeProvider for its clock. Make methods to get the clock and nettimeprovider Add a x-gst-clock property to the SDP with the IP and port number of the nettime provider and also the current time of the clock. This should make it possible for (GStreamer) clients to slave their clock to the server clock.
2013-04-09 20:35:28 +00:00
gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add GstNetTimeProvider support Add a property to let the media provide a GstNetTimeProvider for its clock. Make methods to get the clock and nettimeprovider Add a x-gst-clock property to the SDP with the IP and port number of the nettime provider and also the current time of the clock. This should make it possible for (GStreamer) clients to slave their clock to the server clock.
2013-04-09 20:35:28 +00:00
GstNetTimeProvider * gst_rtsp_media_get_time_provider (GstRTSPMedia *media,
const gchar *address, guint16 port);
media: add property for multicast group Add a property to configure the multicast group in the media. Based on patches from Marc Leeman and Robert Krakora.
2011-08-16 11:25:16 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media: Add API to directly configure a clock on the media pipelines
2015-12-30 14:31:13 +00:00
void gst_rtsp_media_set_clock (GstRTSPMedia *media, GstClock * clock);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_set_publish_clock_mode (GstRTSPMedia * media, GstRTSPPublishClockMode mode);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_get_publish_clock_mode (GstRTSPMedia * media);
Add new API for setting/getting maximum multicast ttl value Change-Id: I5ef4758188c14785e17fb8fbf42a3dc0cb054233 https://bugzilla.gnome.org/show_bug.cgi?id=793441
2018-07-24 07:35:46 +00:00
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia *media, guint ttl);
GST_RTSP_SERVER_API
guint gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia *media);
New property for socket binding to mcast addresses By default the multicast sockets are bound to INADDR_ANY, as it's not allowed to bind sockets to multicast addresses in Windows. This default behaviour can be changed by setting bind-mcast-address property on the media-factory object. https://bugzilla.gnome.org/show_bug.cgi?id=797059
2018-09-06 14:17:33 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia *media, gboolean bind_mcast_addr);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia *media);
media-factory: complete DSCP QoS setting support add dscp_qos setting support at factory and media level to setup IP DSCP field of bounded UDP sinks. Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/6 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/120>
2015-03-03 13:42:07 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_media_set_dscp_qos (GstRTSPMedia * media, gint dscp_qos);
GST_RTSP_SERVER_API
gint gst_rtsp_media_get_dscp_qos (GstRTSPMedia * media);
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
/* prepare the media for playback */
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: Accept a thread in _prepare Remove out own threadpool handling and use the provided thread and maincontext for the bus messages and the state changes.
2013-07-10 15:08:14 +00:00
gboolean gst_rtsp_media_prepare (GstRTSPMedia *media, GstRTSPThread *thread);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: more work on making the media shared Add a reusable flag to medias, indicating that they can be reused after a state change to NULL. Small cleanups.
2009-04-03 20:22:30 +00:00
gboolean gst_rtsp_media_unprepare (GstRTSPMedia *media);
Reorganize things, prepare for media sharing Added various other test server examples Move the SDP message generation to a separate helper. Refactor common code for finding the session. Add content-base for realplayer compatibility Clean up request uris before processing for better vlc compatibility. Move prerolling and pipeline construction to the RTSPMedia object. Use multiudpsink for future pipeline reuse.
2009-01-29 12:31:27 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
void gst_rtsp_media_set_suspend_mode (GstRTSPMedia *media, GstRTSPSuspendMode mode);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add suspend modes Add support for different suspend modes. The stream is suspended right after producing the SDP and after PAUSE. Different suspend modes are available that affect the state of the pipeline. NONE leaves the pipeline state unchanged and is the current and old behaviour, PAUSE will set the pipeline to the PAUSED state and RESET will bring the pipeline to the NULL state. A stream is also unsuspended when it goes back to PLAYING, for RESET streams, this means that the pipeline needs to be prerolled again. Base on patches by Ognyan Tonchev <ognyan@axis.com> See https://bugzilla.gnome.org/show_bug.cgi?id=711257
2013-11-28 13:10:19 +00:00
gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add setup_sdp vmethod gst/rtsp-server/rtsp-media.[ch]: added setup_sdp vmethod and public gst_rtsp_media_setup_sdp. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720155
2013-12-18 15:37:27 +00:00
gboolean gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
GstSDPInfo * info);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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
gboolean gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
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
/* creating streams */
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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
void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_create_stream (GstRTSPMedia *media,
GstElement *payloader,
rtsp-media: Fix small typo causing gtk-doc to complain
2015-08-31 12:47:45 +00:00
GstPad *pad);
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
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
/* dealing with the media */
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-client: Lock shared media For shared media we got race conditions. Concurrently rtsp clients might suspend or unsuspend the shared media and thus change the state without the clients expecting that. By introducing a lock that can be taken by callers such as rtsp_client one can force rtsp clients calling, eg. PLAY, SETUP and that uses shared media, to handle the media sequentially thus allowing one client to finish its rtsp call before another client calls on the same media. https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/86 Fixes #86
2019-10-16 13:20:54 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_media_lock (GstRTSPMedia *media);
GST_RTSP_SERVER_API
void gst_rtsp_media_unlock (GstRTSPMedia *media);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add GstNetTimeProvider support Add a property to let the media provide a GstNetTimeProvider for its clock. Make methods to get the clock and nettimeprovider Add a x-gst-clock property to the SDP with the IP and port number of the nettime provider and also the current time of the clock. This should make it possible for (GStreamer) clients to slave their clock to the server clock.
2013-04-09 20:35:28 +00:00
GstClock * gst_rtsp_media_get_clock (GstRTSPMedia *media);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add method to get the base_time of the pipeline Together with a shared clock, this base-time could eventually be sent to the client so that it can reconstruct the exact running-time of the clock on the server.
2013-04-12 10:34:38 +00:00
GstClockTime gst_rtsp_media_get_base_time (GstRTSPMedia *media);
media: add GstNetTimeProvider support Add a property to let the media provide a GstNetTimeProvider for its clock. Make methods to get the clock and nettimeprovider Add a x-gst-clock property to the SDP with the IP and port number of the nettime provider and also the current time of the clock. This should make it possible for (GStreamer) clients to slave their clock to the server clock.
2013-04-09 20:35:28 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
guint gst_rtsp_media_n_streams (GstRTSPMedia *media);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_get_stream (GstRTSPMedia *media, guint idx);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
media: add method to find a stream by control url
2013-07-03 13:14:39 +00:00
GstRTSPStream * gst_rtsp_media_find_stream (GstRTSPMedia *media, const gchar * control);
Add shared properties to media and factory Add the shared property to media. Implement some simple caching in the factory depending on if the media is shared or not.
2009-01-29 16:20:27 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
Add beginnings of seeking. Parse the Range header and perform a seek on the pipeline for the requested position. It's disabled currently until I figure out what's going wrong.
2009-03-12 19:32:14 +00:00
gboolean gst_rtsp_media_seek (GstRTSPMedia *media, GstRTSPTimeRange *range);
rtsp-server: add missing GST_EXPORT and export deprecated funcs
2017-11-26 12:47:08 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
Start support for RTSP 2.0 This adds basic support for new 2.0 features, though the protocol is subposdely backward incompatible, most semantics are the sames. This commit adds: - features: * version negotiation * pipelined requests support * Media-Properties support * Accept-Ranges support - APIs: * gst_rtsp_media_seekable The RTSP methods that have been removed when using 2.0 now return BAD_REQUEST. https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-04-22 12:26:07 +00:00
gboolean gst_rtsp_media_seek_full (GstRTSPMedia *media,
GstRTSPTimeRange *range,
GstSeekFlags flags);
rtsp-media: allow specifying rate when seeking Add new function gst_rtsp_media_seek_full_with_rate() which allows the caller to specify the rate for the seek. Also added functions in rtsp-stream and rtsp-media for retreiving current rate and applied rate. https://bugzilla.gnome.org/show_bug.cgi?id=754575
2018-09-27 17:09:01 +00:00
GST_RTSP_SERVER_API
onvif: Implement and test the Streaming Specification https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2019-04-04 22:48:07 +00:00
gboolean gst_rtsp_media_seek_trickmode (GstRTSPMedia *media,
GstRTSPTimeRange *range,
GstSeekFlags flags,
gdouble rate,
GstClockTime trickmode_interval);
rtsp-media: allow specifying rate when seeking Add new function gst_rtsp_media_seek_full_with_rate() which allows the caller to specify the rate for the seek. Also added functions in rtsp-stream and rtsp-media for retreiving current rate and applied rate. https://bugzilla.gnome.org/show_bug.cgi?id=754575
2018-09-27 17:09:01 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
Start support for RTSP 2.0 This adds basic support for new 2.0 features, though the protocol is subposdely backward incompatible, most semantics are the sames. This commit adds: - features: * version negotiation * pipelined requests support * Media-Properties support * Accept-Ranges support - APIs: * gst_rtsp_media_seekable The RTSP methods that have been removed when using 2.0 now return BAD_REQUEST. https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-04-22 12:26:07 +00:00
GstClockTimeDiff gst_rtsp_media_seekable (GstRTSPMedia *media);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
rtsp-media/client: Reply to PLAY request with same type of Range Remember the type of Range from the PLAY request and use the same type for the reply.
2013-02-22 19:17:29 +00:00
gchar * gst_rtsp_media_get_range_string (GstRTSPMedia *media,
gboolean play,
GstRTSPRangeUnit unit);
Add beginnings of seeking. Parse the Range header and perform a seek on the pipeline for the requested position. It's disabled currently until I figure out what's going wrong.
2009-03-12 19:32:14 +00:00
rtsp-media: allow specifying rate when seeking Add new function gst_rtsp_media_seek_full_with_rate() which allows the caller to specify the rate for the seek. Also added functions in rtsp-stream and rtsp-media for retreiving current rate and applied rate. https://bugzilla.gnome.org/show_bug.cgi?id=754575
2018-09-27 17:09:01 +00:00
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_rates (GstRTSPMedia * media,
gdouble * rate,
gdouble * applied_rate);
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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_media_set_state (GstRTSPMedia *media, GstState state,
GPtrArray *transports);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
Add API to rtsp-media set the pipeline's state
2013-07-02 22:55:28 +00:00
void gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media,
GstState state);
Added gst_rtsp_media_remove_elements function
2009-06-13 12:38:20 +00:00
rtsp-server: GST_EXPORT -> GST_RTSP_SERVER_API We need different export decorators for the different libs. For now no actual change though, just rename before the release, and add prelude headers to define the new decorator to GST_EXPORT.
2018-03-13 13:30:41 +00:00
GST_RTSP_SERVER_API
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
gboolean gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports);
rtsp-client: add support for Scale and Speed header Add support for the RTSP Scale and Speed headers by setting the rate in the seek to (scale*speed). We then check the resulting segment for rate and applied rate, and use them as values for the Speed and Scale headers respectively. https://bugzilla.gnome.org/show_bug.cgi?id=754575
2018-10-12 06:53:04 +00:00
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_is_receive_only (GstRTSPMedia * media);
Use complete streams for scale and speed. Without this patch it's always stream0 that is used to get segment event that is used to set scale and speed. This even if client not doing SETUP for stream0. At least in suspend mode reset this not working since then it's just random if send_rtp_sink have got any segment event. There are no check if send_rtp_sink for stream0 got any data before media is prerolled after PLAY request.
2019-08-21 12:57:25 +00:00
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_has_completed_sender (GstRTSPMedia * media);
onvif: Implement and test the Streaming Specification https://www.onvif.org/specs/stream/ONVIF-Streaming-Spec.pdf
2019-04-04 22:48:07 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled);
GST_RTSP_SERVER_API
gboolean gst_rtsp_media_get_rate_control (GstRTSPMedia * media);
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(GstRTSPMedia, gst_object_unref)
#endif
Initial import
2008-10-09 12:29:12 +00:00
G_END_DECLS
#endif /* __GST_RTSP_MEDIA_H__ */
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