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

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rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
/* GStreamer
* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
Fix FSF address
2012-11-04 00:14:25 +00:00
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
*/
#include <gst/gst.h>
stream: refactor TCP backpressure handling The previous implementation stopped sending TCP messages to all clients when a single one stopped consuming them, which obviously created problems for shared media. Instead, we now manage a backlog in stream-transport, and slow clients are removed once this backlog exceeds a maximum duration, currently hardcoded. Fixes #80
2019-10-15 17:08:32 +00:00
#include <gst/base/base.h>
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
#include <gst/rtsp/gstrtsprange.h>
#include <gst/rtsp/gstrtspurl.h>
#ifndef __GST_RTSP_STREAM_TRANSPORT_H__
#define __GST_RTSP_STREAM_TRANSPORT_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"
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
G_BEGIN_DECLS
/* types for the media */
#define GST_TYPE_RTSP_STREAM_TRANSPORT (gst_rtsp_stream_transport_get_type ())
#define GST_IS_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT))
#define GST_IS_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT))
#define GST_RTSP_STREAM_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
#define GST_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransport))
#define GST_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
#define GST_RTSP_STREAM_TRANSPORT_CAST(obj) ((GstRTSPStreamTransport*)(obj))
#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
typedef struct _GstRTSPStreamTransport GstRTSPStreamTransport;
typedef struct _GstRTSPStreamTransportClass GstRTSPStreamTransportClass;
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 _GstRTSPStreamTransportPrivate GstRTSPStreamTransportPrivate;
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
#include "rtsp-stream.h"
docs: update docs
2013-07-11 10:18:26 +00:00
/**
* GstRTSPSendFunc:
* @buffer: a #GstBuffer
* @channel: a channel
* @user_data: user data
*
* Function registered with gst_rtsp_stream_transport_set_callbacks() and
* called when @buffer must be sent on @channel.
*
* Returns: %TRUE on success
*/
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
typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
rtsp-server: Add support for buffer lists This adds new functions for passing buffer lists through the different layers without breaking API/ABI, and enables the appsink to actually provide buffer lists. This should already reduce CPU usage and potentially context switches a bit by passing a whole buffer list from the appsink instead of individual buffers. As a next step it would be necessary to a) Add support for a vector of data for the GstRTSPMessage body b) Add support for sending multiple messages at once to the GstRTSPWatch and let it be handled internally c) Adding API to GOutputStream that works like writev() Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2018-06-27 10:17:07 +00:00
/**
* GstRTSPSendListFunc:
* @buffer_list: a #GstBufferList
* @channel: a channel
* @user_data: user data
*
* Function registered with gst_rtsp_stream_transport_set_callbacks() and
* called when @buffer_list must be sent on @channel.
*
* Returns: %TRUE on success
*
* Since: 1.16
*/
typedef gboolean (*GstRTSPSendListFunc) (GstBufferList *buffer_list, guint8 channel, gpointer user_data);
docs: update docs
2013-07-11 10:18:26 +00:00
/**
* GstRTSPKeepAliveFunc:
* @user_data: user data
*
* Function registered with gst_rtsp_stream_transport_set_keepalive() and called
* when the stream is active.
*/
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
typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
Limit queued TCP data messages to one per stream Before, the watch backlog size in GstRTSPClient was changed dynamically between unlimited and a fixed size, trying to avoid both unlimited memory usage and deadlocks while waiting for place in the queue. (Some of the deadlocks were described in a long comment in handle_request().) In the previous commit, we changed to a fixed backlog size of 100. This is possible, because we now handle RTP/RTCP data messages differently from RTSP request/response messages. The data messages are messages tunneled over TCP. We allow at most one queued data message per stream in GstRTSPClient at a time, and successfully sent data messages are acked by sending a "message-sent" callback from the GstStreamTransport. Until that ack comes, the GstRTSPStream does not call pull_sample() on its appsink, and therefore the streaming thread in the pipeline will not be blocked inside GstRTSPClient, waiting for a place in the queue. pull_sample() is called when we have both an ack and a "new-sample" signal from the appsink. Then, we know there is a buffer to write. RTSP request/response messages are not acked in the same way as data messages. The rest of the 100 places in the queue are used for them. If the queue becomes full of request/response messages, we return an error and close the connection to the client. Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-06-28 09:22:21 +00:00
/**
* GstRTSPMessageSentFunc:
* @user_data: user data
*
* Function registered with gst_rtsp_stream_transport_set_message_sent()
* and called when a message has been sent on the transport.
*/
rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks The old API is preserved now and new API was added that provides the additional parameter to the callback. Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
2020-05-01 07:42:17 +00:00
typedef void (*GstRTSPMessageSentFunc) (gpointer user_data);
/**
* GstRTSPMessageSentFuncFull:
* @user_data: user data
*
* Function registered with gst_rtsp_stream_transport_set_message_sent_full()
* and called when a message has been sent on the transport.
*
* Since: 1.18
*/
typedef void (*GstRTSPMessageSentFuncFull) (GstRTSPStreamTransport *trans, gpointer user_data);
Limit queued TCP data messages to one per stream Before, the watch backlog size in GstRTSPClient was changed dynamically between unlimited and a fixed size, trying to avoid both unlimited memory usage and deadlocks while waiting for place in the queue. (Some of the deadlocks were described in a long comment in handle_request().) In the previous commit, we changed to a fixed backlog size of 100. This is possible, because we now handle RTP/RTCP data messages differently from RTSP request/response messages. The data messages are messages tunneled over TCP. We allow at most one queued data message per stream in GstRTSPClient at a time, and successfully sent data messages are acked by sending a "message-sent" callback from the GstStreamTransport. Until that ack comes, the GstRTSPStream does not call pull_sample() on its appsink, and therefore the streaming thread in the pipeline will not be blocked inside GstRTSPClient, waiting for a place in the queue. pull_sample() is called when we have both an ack and a "new-sample" signal from the appsink. Then, we know there is a buffer to write. RTSP request/response messages are not acked in the same way as data messages. The rest of the 100 places in the queue are used for them. If the queue becomes full of request/response messages, we return an error and close the connection to the client. Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-06-28 09:22:21 +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
/**
* GstRTSPStreamTransport:
* @parent: parent instance
*
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
* A Transport description for a stream
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
*/
struct _GstRTSPStreamTransport {
GObject parent;
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
GstRTSPStreamTransportPrivate *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];
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
};
struct _GstRTSPStreamTransportClass {
GObjectClass parent_class;
rtsp-server: add padding to many public structures Not mini objects though, since they are not subclassable anyway, nor kept on the stack or inlined in a structure.
2013-12-12 00:36:07 +00:00
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
};
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
GType gst_rtsp_stream_transport_get_type (void);
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: only create transport when needed Only create the StreamTransport when configured.
2012-10-27 22:23:57 +00:00
GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream,
stream: use the address managed by the stream Use the address managed by the stream for multicast. This allows us to have 1 multicast address for each stream. Because the address is now managed by the stream we don't have to pass it around anymore. Set the address pool on the streams.
2012-11-15 15:18:29 +00:00
GstRTSPTransport *tr);
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
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
GstRTSPStream * gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport *trans);
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: refactor configuration of transport Move the configuration of the transport to a place where it makes more sense.
2012-10-27 21:49:24 +00:00
void gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
stream: use the address managed by the stream Use the address managed by the stream for multicast. This allows us to have 1 multicast address for each stream. Because the address is now managed by the stream we don't have to pass it around anymore. Set the address pool on the streams.
2012-11-15 15:18:29 +00:00
GstRTSPTransport * tr);
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
const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport *trans);
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
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-transport: add method to get/set url
2013-11-28 16:35:45 +00:00
void gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport *trans,
const GstRTSPUrl * url);
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-transport: add method to get/set url
2013-11-28 16:35:45 +00:00
const GstRTSPUrl * gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport *trans);
session-media: let the session-media make the RTPInfo Add method to create the RTPInfo for a stream-transport. Add method to create the RTPInfo for all stream-transports in a session-media. Use the session-media RTPInfo code in client. This allows us to refactor another method to link the TCP callbacks.
2013-12-26 14:41:14 +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
session-media: calculate start-time
2013-12-26 15:28:59 +00:00
gchar * gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport *trans,
GstClockTime start_time);
session-media: let the session-media make the RTPInfo Add method to create the RTPInfo for a stream-transport. Add method to create the RTPInfo for all stream-transports in a session-media. Use the session-media RTPInfo code in client. This allows us to refactor another method to link the TCP callbacks.
2013-12-26 14:41:14 +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_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
GstRTSPSendFunc send_rtp,
GstRTSPSendFunc send_rtcp,
gpointer user_data,
GDestroyNotify notify);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
rtsp-server: Add support for buffer lists This adds new functions for passing buffer lists through the different layers without breaking API/ABI, and enables the appsink to actually provide buffer lists. This should already reduce CPU usage and potentially context switches a bit by passing a whole buffer list from the appsink instead of individual buffers. As a next step it would be necessary to a) Add support for a vector of data for the GstRTSPMessage body b) Add support for sending multiple messages at once to the GstRTSPWatch and let it be handled internally c) Adding API to GOutputStream that works like writev() Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2018-06-27 10:17:07 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_stream_transport_set_list_callbacks (GstRTSPStreamTransport *trans,
GstRTSPSendListFunc send_rtp_list,
GstRTSPSendListFunc send_rtcp_list,
gpointer user_data,
GDestroyNotify notify);
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_stream_transport_set_keepalive (GstRTSPStreamTransport *trans,
GstRTSPKeepAliveFunc keep_alive,
gpointer user_data,
GDestroyNotify notify);
Mark symbols explicitly for export with GST_EXPORT
2017-05-18 09:35:18 +00:00
Limit queued TCP data messages to one per stream Before, the watch backlog size in GstRTSPClient was changed dynamically between unlimited and a fixed size, trying to avoid both unlimited memory usage and deadlocks while waiting for place in the queue. (Some of the deadlocks were described in a long comment in handle_request().) In the previous commit, we changed to a fixed backlog size of 100. This is possible, because we now handle RTP/RTCP data messages differently from RTSP request/response messages. The data messages are messages tunneled over TCP. We allow at most one queued data message per stream in GstRTSPClient at a time, and successfully sent data messages are acked by sending a "message-sent" callback from the GstStreamTransport. Until that ack comes, the GstRTSPStream does not call pull_sample() on its appsink, and therefore the streaming thread in the pipeline will not be blocked inside GstRTSPClient, waiting for a place in the queue. pull_sample() is called when we have both an ack and a "new-sample" signal from the appsink. Then, we know there is a buffer to write. RTSP request/response messages are not acked in the same way as data messages. The rest of the 100 places in the queue are used for them. If the queue becomes full of request/response messages, we return an error and close the connection to the client. Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-06-28 09:22:21 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport *trans,
rtsp-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks The old API is preserved now and new API was added that provides the additional parameter to the callback. Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
2020-05-01 07:42:17 +00:00
GstRTSPMessageSentFunc message_sent,
gpointer user_data,
GDestroyNotify notify);
Limit queued TCP data messages to one per stream Before, the watch backlog size in GstRTSPClient was changed dynamically between unlimited and a fixed size, trying to avoid both unlimited memory usage and deadlocks while waiting for place in the queue. (Some of the deadlocks were described in a long comment in handle_request().) In the previous commit, we changed to a fixed backlog size of 100. This is possible, because we now handle RTP/RTCP data messages differently from RTSP request/response messages. The data messages are messages tunneled over TCP. We allow at most one queued data message per stream in GstRTSPClient at a time, and successfully sent data messages are acked by sending a "message-sent" callback from the GstStreamTransport. Until that ack comes, the GstRTSPStream does not call pull_sample() on its appsink, and therefore the streaming thread in the pipeline will not be blocked inside GstRTSPClient, waiting for a place in the queue. pull_sample() is called when we have both an ack and a "new-sample" signal from the appsink. Then, we know there is a buffer to write. RTSP request/response messages are not acked in the same way as data messages. The rest of the 100 places in the queue are used for them. If the queue becomes full of request/response messages, we return an error and close the connection to the client. Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-06-28 09:22:21 +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-stream-transport: Fix accidental API/ABI breakage with message_sent callbacks The old API is preserved now and new API was added that provides the additional parameter to the callback. Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/issues/104 Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/-/merge_requests/116>
2020-05-01 07:42:17 +00:00
void gst_rtsp_stream_transport_set_message_sent_full (GstRTSPStreamTransport *trans,
GstRTSPMessageSentFuncFull message_sent,
gpointer user_data,
GDestroyNotify notify);
GST_RTSP_SERVER_API
docs: improve docs
2013-07-16 10:32:51 +00:00
void gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport *trans);
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
Limit queued TCP data messages to one per stream Before, the watch backlog size in GstRTSPClient was changed dynamically between unlimited and a fixed size, trying to avoid both unlimited memory usage and deadlocks while waiting for place in the queue. (Some of the deadlocks were described in a long comment in handle_request().) In the previous commit, we changed to a fixed backlog size of 100. This is possible, because we now handle RTP/RTCP data messages differently from RTSP request/response messages. The data messages are messages tunneled over TCP. We allow at most one queued data message per stream in GstRTSPClient at a time, and successfully sent data messages are acked by sending a "message-sent" callback from the GstStreamTransport. Until that ack comes, the GstRTSPStream does not call pull_sample() on its appsink, and therefore the streaming thread in the pipeline will not be blocked inside GstRTSPClient, waiting for a place in the queue. pull_sample() is called when we have both an ack and a "new-sample" signal from the appsink. Then, we know there is a buffer to write. RTSP request/response messages are not acked in the same way as data messages. The rest of the 100 places in the queue are used for them. If the queue becomes full of request/response messages, we return an error and close the connection to the client. Change-Id: I275310bc90a219ceb2473c098261acc78be84c97
2018-06-28 09:22:21 +00:00
GST_RTSP_SERVER_API
void gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport *trans);
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
gboolean gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport *trans,
gboolean active);
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_stream_transport_set_timed_out (GstRTSPStreamTransport *trans,
gboolean timedout);
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
gboolean gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport *trans);
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-transport: add method to handle RTP/RTCP Call new methods instead of poking into the structures directly.
2012-11-12 16:06:42 +00:00
gboolean gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
GstBuffer *buffer);
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-transport: add method to handle RTP/RTCP Call new methods instead of poking into the structures directly.
2012-11-12 16:06:42 +00:00
gboolean gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
GstBuffer *buffer);
rtsp-server: Add support for buffer lists This adds new functions for passing buffer lists through the different layers without breaking API/ABI, and enables the appsink to actually provide buffer lists. This should already reduce CPU usage and potentially context switches a bit by passing a whole buffer list from the appsink instead of individual buffers. As a next step it would be necessary to a) Add support for a vector of data for the GstRTSPMessage body b) Add support for sending multiple messages at once to the GstRTSPWatch and let it be handled internally c) Adding API to GOutputStream that works like writev() Fixes https://gitlab.freedesktop.org/gstreamer/gst-rtsp-server/issues/29
2018-06-27 10:17:07 +00:00
GST_RTSP_SERVER_API
gboolean gst_rtsp_stream_transport_send_rtp_list (GstRTSPStreamTransport *trans,
GstBufferList *buffer_list);
GST_RTSP_SERVER_API
gboolean gst_rtsp_stream_transport_send_rtcp_list(GstRTSPStreamTransport *trans,
GstBufferList *buffer_list);
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-transport: make method to handle received data Make a method to handle the data received on a channel. It sends the data to the stream of the transport on the RTP or RTCP pads based on the channel number.
2014-09-09 16:10:12 +00:00
GstFlowReturn gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport *trans,
guint channel, GstBuffer *buffer);
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(GstRTSPStreamTransport, gst_object_unref)
#endif
rtsp: massive refactoring Make GObjects from the remaining simple structures. Remove GstRTSPSessionStream, it's not needed. Rename GstRTSPMediaStream -> GstRTSPStream: It is shorter Rename GstRTSPMediaTrans -> GstRTSPStreamTransport: It describes how a GstRTSPStream should be transported to a client. Rename GstRTSPMediaFactory::get_element -> create_element because that more accurately describes what it does. Make nice methods instead of poking in the structures. Move some methods inside the relevant object source code. Use GPtrArray to store objects instead of plain arrays, it is more natural and allows us to more easily clean up. Move the allocation of udp ports to the Stream object. The Stream object contains the elements needed to stream the media to a client. Improve the prepare and unprepare methods. Unprepare should now undo everything prepare did. Improve also async unprepare when doing EOS on shutdown. Make sure we always unprepare correctly.
2012-10-25 19:29:58 +00:00
G_END_DECLS
#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */
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