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gstreamer/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>
Cleanup of sessions and more Fix the refcounting of media and sessions in the client. Properly clean up the session data when the client performs a teardown. Add Server header to responses. Allow for multiple uri setups in one session. Add Range header to the PLAY response and add the range attribute to the SDP message. Fix the session pool remove method, it used the wrong key in the hashtable. Also give the ownership of the sessionid to the session object.
2009-02-04 16:00:42 +00:00
#include <gst/rtsp/gstrtsprange.h>
Initial import
2008-10-09 12:29:12 +00:00
#include <gst/rtsp/gstrtspurl.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__
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
* after creating the SDP and when the client preforms a PAUSED request.
*/
typedef enum {
GST_RTSP_SUSPEND_MODE_NONE = 0,
GST_RTSP_SUSPEND_MODE_PAUSE = 1,
GST_RTSP_SUSPEND_MODE_RESET = 2
} GstRTSPSuspendMode;
#define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
GType gst_rtsp_suspend_mode_get_type (void);
#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
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.
docs: more updates
2013-07-11 10:24:33 +00:00
* @convert_range: convert a range to the given unit
* @query_position: query the current posision in the pipeline
* @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);
gboolean (*unprepare) (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: 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
/*< private >*/
gpointer _gst_reserved[GST_PADDING_LARGE];
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
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 */
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);
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
void gst_rtsp_media_take_pipeline (GstRTSPMedia *media, GstPipeline *pipeline);
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
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);
GstRTSPPermissions * gst_rtsp_media_get_permissions (GstRTSPMedia *media);
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);
gboolean gst_rtsp_media_is_shared (GstRTSPMedia *media);
Initial import
2008-10-09 12:29:12 +00:00
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);
gboolean gst_rtsp_media_is_reusable (GstRTSPMedia *media);
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);
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
void gst_rtsp_media_set_eos_shutdown (GstRTSPMedia *media, gboolean eos_shutdown);
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: 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);
GstRTSPAddressPool * gst_rtsp_media_get_address_pool (GstRTSPMedia *media);
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);
guint gst_rtsp_media_get_buffer_size (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
void gst_rtsp_media_use_time_provider (GstRTSPMedia *media, gboolean time_provider);
gboolean gst_rtsp_media_is_time_provider (GstRTSPMedia *media);
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
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 */
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);
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
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);
GstRTSPSuspendMode gst_rtsp_media_get_suspend_mode (GstRTSPMedia *media);
gboolean gst_rtsp_media_suspend (GstRTSPMedia *media);
gboolean gst_rtsp_media_unsuspend (GstRTSPMedia *media);
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: 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 */
void gst_rtsp_media_collect_streams (GstRTSPMedia *media);
GstRTSPStream * gst_rtsp_media_create_stream (GstRTSPMedia *media,
GstElement *payloader,
GstPad *srcpad);
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 */
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);
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
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);
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);
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
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-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: 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);
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
Initial import
2008-10-09 12:29:12 +00:00
G_END_DECLS
#endif /* __GST_RTSP_MEDIA_H__ */
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