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

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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
*/
docs: improve docs
2013-07-11 14:57:14 +00:00
/**
* SECTION:rtsp-stream-transport
* @short_description: A media stream transport configuration
docs: improve docs
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* @see_also: #GstRTSPStream, #GstRTSPSessionMedia
docs: improve docs
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*
docs: improve docs
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* The #GstRTSPStreamTransport configures the transport used by a
* #GstRTSPStream. It is usually manages by a #GstRTSPSessionMedia object.
*
* With gst_rtsp_stream_transport_set_callbacks(), callbacks can be configured
* to handle the RTP and RTCP packets from the stream, for example when they
* need to be sent over TCP.
*
rtsp-stream: Removing invalid transports returns false When removing transports an assertion was that the transports passed in for removal are present in the list, however that can't be assumed. As an example if a transport was removed from a thread running send_tcp_message, the main thread can try to remove the same transport again if it gets a handle_pause_request. This will not effect the transport list but it will effect n_tcp_transports as it will be decrement and then have the wrong value.
2019-11-01 11:01:41 +00:00
* With gst_rtsp_stream_transport_set_active() the transports are added and
docs: improve docs
2013-07-16 10:32:51 +00:00
* removed from the stream.
*
* A #GstRTSPStream will call gst_rtsp_stream_transport_keep_alive() when RTCP
* is received from the client. It will also call
* gst_rtsp_stream_transport_set_timed_out() when a receiver has timed out.
*
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
* A #GstRTSPClient will call gst_rtsp_stream_transport_message_sent() when it
* has sent a data message for the transport.
*
docs: improve docs
2013-07-16 10:32:51 +00:00
* Last reviewed on 2013-07-16 (1.0.0)
docs: improve docs
2013-07-11 14:57:14 +00:00
*/
libs: fix API export/import and 'inconsistent linkage' on MSVC Export rtsp-server library API in headers when we're building the library itself, otherwise import the API from the headers. This fixes linker warnings on Windows when building with MSVC. Fix up some missing config.h includes when building the lib which is needed to get the export api define from config.h https://bugzilla.gnome.org/show_bug.cgi?id=797185
2018-09-24 08:36:21 +00:00
#ifdef HAVE_CONFIG_H
#include "config.h"
#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
#include <string.h>
#include <stdlib.h>
#include "rtsp-stream-transport.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 "rtsp-server-internal.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
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
struct _GstRTSPStreamTransportPrivate
{
GstRTSPStream *stream;
GstRTSPSendFunc send_rtp;
GstRTSPSendFunc send_rtcp;
gpointer user_data;
GDestroyNotify notify;
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 send_rtp_list;
GstRTSPSendListFunc send_rtcp_list;
gpointer list_user_data;
GDestroyNotify list_notify;
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
GstRTSPBackPressureFunc back_pressure_func;
gpointer back_pressure_func_data;
GDestroyNotify back_pressure_func_notify;
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
GstRTSPKeepAliveFunc keep_alive;
gpointer ka_user_data;
GDestroyNotify ka_notify;
gboolean timed_out;
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 message_sent;
gpointer ms_user_data;
GDestroyNotify ms_notify;
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
GstRTSPMessageSentFuncFull message_sent_full;
gpointer msf_user_data;
GDestroyNotify msf_notify;
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
GstRTSPTransport *transport;
stream-transport: add method to get/set url
2013-11-28 16:35:45 +00:00
GstRTSPUrl *url;
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
GObject *rtpsource;
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
/* TCP backlog */
GstClockTime first_rtp_timestamp;
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
GstVecDeque *items;
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
GRecMutex backlog_lock;
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
};
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
#define MAX_BACKLOG_DURATION (10 * GST_SECOND)
#define MAX_BACKLOG_SIZE 100
typedef struct
{
GstBuffer *buffer;
GstBufferList *buffer_list;
gboolean is_rtp;
} BackLogItem;
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
enum
{
PROP_0,
PROP_LAST
};
GST_DEBUG_CATEGORY_STATIC (rtsp_stream_transport_debug);
#define GST_CAT_DEFAULT rtsp_stream_transport_debug
static void gst_rtsp_stream_transport_finalize (GObject * obj);
Update for g_type_class_add_private() deprecation in recent GLib
2018-06-24 10:44:26 +00:00
G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPStreamTransport, gst_rtsp_stream_transport,
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_TYPE_OBJECT);
static void
gst_rtsp_stream_transport_class_init (GstRTSPStreamTransportClass * klass)
{
GObjectClass *gobject_class;
gobject_class = G_OBJECT_CLASS (klass);
gobject_class->finalize = gst_rtsp_stream_transport_finalize;
GST_DEBUG_CATEGORY_INIT (rtsp_stream_transport_debug, "rtspmediatransport",
0, "GstRTSPStreamTransport");
}
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
static void
clear_backlog_item (BackLogItem * item)
{
gst_clear_buffer (&item->buffer);
gst_clear_buffer_list (&item->buffer_list);
}
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
static void
gst_rtsp_stream_transport_init (GstRTSPStreamTransport * trans)
{
Update for g_type_class_add_private() deprecation in recent GLib
2018-06-24 10:44:26 +00:00
trans->priv = gst_rtsp_stream_transport_get_instance_private (trans);
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
trans->priv->items = gst_vec_deque_new_for_struct (sizeof (BackLogItem), 0);
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
trans->priv->first_rtp_timestamp = GST_CLOCK_TIME_NONE;
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
gst_vec_deque_set_clear_func (trans->priv->items,
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
(GDestroyNotify) clear_backlog_item);
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
g_rec_mutex_init (&trans->priv->backlog_lock);
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
}
static void
gst_rtsp_stream_transport_finalize (GObject * obj)
{
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: 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 *trans;
trans = GST_RTSP_STREAM_TRANSPORT (obj);
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv = trans->priv;
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
/* remove callbacks now */
gst_rtsp_stream_transport_set_callbacks (trans, NULL, NULL, NULL, NULL);
gst_rtsp_stream_transport_set_keepalive (trans, NULL, NULL, NULL);
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_stream_transport_set_message_sent (trans, NULL, NULL, NULL);
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: Ref transports in hash table. Also ref streams for transports. This solves a crash when reciving a rtcp after teardown but before client finalize. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2014-11-28 11:45:14 +00:00
if (priv->stream)
g_object_unref (priv->stream);
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
if (priv->transport)
gst_rtsp_transport_free (priv->transport);
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
stream-transport: free url in finalize
2013-11-29 14:50:52 +00:00
if (priv->url)
gst_rtsp_url_free (priv->url);
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
gst_vec_deque_free (priv->items);
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
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
g_rec_mutex_clear (&priv->backlog_lock);
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_OBJECT_CLASS (gst_rtsp_stream_transport_parent_class)->finalize (obj);
}
/**
* gst_rtsp_stream_transport_new:
* @stream: a #GstRTSPStream
rtsp: only create transport when needed Only create the StreamTransport when configured.
2012-10-27 22:23:57 +00:00
* @tr: (transfer full): a GstRTSPTransport
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: 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
* Create a new #GstRTSPStreamTransport that can be used to manage
rtsp: only create transport when needed Only create the StreamTransport when configured.
2012-10-27 22:23:57 +00:00
* @stream with transport @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
*
gobject-introspection: Add annotations to support language bindings In addition a few cosmetic changes: * Adjust the order of arguments * Fix typo: occured -> occurred * Fix indentation after Return:-clauses Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-23 20:24:48 +00:00
* Returns: (transfer full): a new #GstRTSPStreamTransport
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 *
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
gst_rtsp_stream_transport_new (GstRTSPStream * stream, 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: 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: 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 *trans;
g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
rtsp: only create transport when needed Only create the StreamTransport when configured.
2012-10-27 22:23:57 +00:00
g_return_val_if_fail (tr != NULL, NULL);
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
trans = g_object_new (GST_TYPE_RTSP_STREAM_TRANSPORT, NULL);
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv = trans->priv;
priv->stream = stream;
rtsp: Ref transports in hash table. Also ref streams for transports. This solves a crash when reciving a rtcp after teardown but before client finalize. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740845
2014-11-28 11:45:14 +00:00
priv->stream = g_object_ref (priv->stream);
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv->transport = 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
return trans;
}
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
/**
* gst_rtsp_stream_transport_get_stream:
* @trans: a #GstRTSPStreamTransport
*
* Get the #GstRTSPStream used when constructing @trans.
*
All around: add annotations and API guards
2018-02-09 23:07:25 +00:00
* Returns: (transfer none) (nullable): the stream used when constructing @trans.
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)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
return trans->priv->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
/**
* gst_rtsp_stream_transport_set_callbacks:
* @trans: a #GstRTSPStreamTransport
all: Fix closure annotations This was misused almost everywhere. See https://gi.readthedocs.io/en/latest/annotations/giannotations.html#support-for-gobject-closures and: https://www.bassi.io/articles/2023/02/20/bindable-api-2023/ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7725>
2024-10-24 07:17:54 +00:00
* @send_rtp: (scope notified) (closure user_data): a callback called when RTP should be sent
* @send_rtcp: (scope notified) (closure user_data): a callback called when RTCP should be sent
* @user_data: user data passed to callbacks
gobject-introspection: Add annotations to support language bindings In addition a few cosmetic changes: * Adjust the order of arguments * Fix typo: occured -> occurred * Fix indentation after Return:-clauses Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-23 20:24:48 +00:00
* @notify: (allow-none): called with the user_data when no longer needed.
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
*
* Install callbacks that will be called when data for a stream should be sent
* to a client. This is usually used when sending RTP/RTCP over TCP.
*/
void
gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans,
GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp,
gpointer user_data, GDestroyNotify notify)
{
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;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->send_rtp = send_rtp;
priv->send_rtcp = send_rtcp;
if (priv->notify)
priv->notify (priv->user_data);
priv->user_data = user_data;
priv->notify = notify;
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: 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_stream_transport_set_list_callbacks:
* @trans: a #GstRTSPStreamTransport
all: Fix closure annotations This was misused almost everywhere. See https://gi.readthedocs.io/en/latest/annotations/giannotations.html#support-for-gobject-closures and: https://www.bassi.io/articles/2023/02/20/bindable-api-2023/ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7725>
2024-10-24 07:17:54 +00:00
* @send_rtp_list: (scope notified) (closure user_data): a callback called when RTP should be sent
* @send_rtcp_list: (scope notified) (closure user_data): a callback called when RTCP should be sent
* @user_data: user data passed to callbacks
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
* @notify: (allow-none): called with the user_data when no longer needed.
*
* Install callbacks that will be called when data for a stream should be sent
* to a client. This is usually used when sending RTP/RTCP over TCP.
*
* Since: 1.16
*/
void
gst_rtsp_stream_transport_set_list_callbacks (GstRTSPStreamTransport * trans,
GstRTSPSendListFunc send_rtp_list, GstRTSPSendListFunc send_rtcp_list,
gpointer user_data, GDestroyNotify notify)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->send_rtp_list = send_rtp_list;
priv->send_rtcp_list = send_rtcp_list;
if (priv->list_notify)
priv->list_notify (priv->list_user_data);
priv->list_user_data = user_data;
priv->list_notify = notify;
}
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
void
gst_rtsp_stream_transport_set_back_pressure_callback (GstRTSPStreamTransport *
trans, GstRTSPBackPressureFunc back_pressure_func, gpointer user_data,
GDestroyNotify notify)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->back_pressure_func = back_pressure_func;
if (priv->back_pressure_func_notify)
priv->back_pressure_func_notify (priv->back_pressure_func_data);
priv->back_pressure_func_data = user_data;
priv->back_pressure_func_notify = notify;
}
gboolean
gst_rtsp_stream_transport_check_back_pressure (GstRTSPStreamTransport * trans,
rtsp-stream: fix checking of TCP backpressure The internal index of our appsinks, while it can be used to determine whether a message is RTP or RTCP, is not necessarily the same as the interleaved channel. Let the stream-transport determine the channel to check backpressure for, the same way it determines the channel according to whether it is sending RTP or RTCP.
2020-01-09 13:10:44 +00:00
gboolean is_rtp)
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
{
GstRTSPStreamTransportPrivate *priv;
gboolean ret = FALSE;
rtsp-stream: fix checking of TCP backpressure The internal index of our appsinks, while it can be used to determine whether a message is RTP or RTCP, is not necessarily the same as the interleaved channel. Let the stream-transport determine the channel to check backpressure for, the same way it determines the channel according to whether it is sending RTP or RTCP.
2020-01-09 13:10:44 +00:00
guint8 channel;
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
priv = trans->priv;
rtsp-stream: fix checking of TCP backpressure The internal index of our appsinks, while it can be used to determine whether a message is RTP or RTCP, is not necessarily the same as the interleaved channel. Let the stream-transport determine the channel to check backpressure for, the same way it determines the channel according to whether it is sending RTP or RTCP.
2020-01-09 13:10:44 +00:00
if (is_rtp)
channel = priv->transport->interleaved.min;
else
channel = priv->transport->interleaved.max;
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
if (priv->back_pressure_func)
ret = priv->back_pressure_func (channel, priv->back_pressure_func_data);
return ret;
}
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_stream_transport_set_keepalive:
* @trans: a #GstRTSPStreamTransport
all: Fix closure annotations This was misused almost everywhere. See https://gi.readthedocs.io/en/latest/annotations/giannotations.html#support-for-gobject-closures and: https://www.bassi.io/articles/2023/02/20/bindable-api-2023/ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7725>
2024-10-24 07:17:54 +00:00
* @keep_alive: (scope notified) (closure user_data): a callback called when the receiver is active
* @user_data: user data passed to callback
gobject-introspection: Add annotations to support language bindings In addition a few cosmetic changes: * Adjust the order of arguments * Fix typo: occured -> occurred * Fix indentation after Return:-clauses Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-23 20:24:48 +00:00
* @notify: (allow-none): called with the user_data when no longer needed.
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
*
* Install callbacks that will be called when RTCP packets are received from the
* receiver of @trans.
*/
void
gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans,
GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
{
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;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->keep_alive = keep_alive;
if (priv->ka_notify)
priv->ka_notify (priv->ka_user_data);
priv->ka_user_data = user_data;
priv->ka_notify = notify;
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_stream_transport_set_message_sent:
* @trans: a #GstRTSPStreamTransport
all: Fix closure annotations This was misused almost everywhere. See https://gi.readthedocs.io/en/latest/annotations/giannotations.html#support-for-gobject-closures and: https://www.bassi.io/articles/2023/02/20/bindable-api-2023/ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7725>
2024-10-24 07:17:54 +00:00
* @message_sent: (scope notified) (closure user_data): a callback called when a message has been sent
* @user_data: user data passed to callback
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
* @notify: (allow-none): called with the user_data when no longer needed
*
* Install a callback that will be called when a message has been sent on @trans.
*/
void
gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport * trans,
GstRTSPMessageSentFunc message_sent, gpointer user_data,
GDestroyNotify notify)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->message_sent = message_sent;
if (priv->ms_notify)
priv->ms_notify (priv->ms_user_data);
priv->ms_user_data = user_data;
priv->ms_notify = notify;
}
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
/**
* gst_rtsp_stream_transport_set_message_sent_full:
* @trans: a #GstRTSPStreamTransport
all: Fix closure annotations This was misused almost everywhere. See https://gi.readthedocs.io/en/latest/annotations/giannotations.html#support-for-gobject-closures and: https://www.bassi.io/articles/2023/02/20/bindable-api-2023/ Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7725>
2024-10-24 07:17:54 +00:00
* @message_sent: (scope notified) (closure user_data): a callback called when a message has been sent
* @user_data: user data passed to callback
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
* @notify: (allow-none): called with the user_data when no longer needed
*
* Install a callback that will be called when a message has been sent on @trans.
*
* Since: 1.18
*/
void
gst_rtsp_stream_transport_set_message_sent_full (GstRTSPStreamTransport * trans,
GstRTSPMessageSentFuncFull message_sent, gpointer user_data,
GDestroyNotify notify)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
priv->message_sent_full = message_sent;
if (priv->msf_notify)
priv->msf_notify (priv->msf_user_data);
priv->msf_user_data = user_data;
priv->msf_notify = notify;
}
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_stream_transport_set_transport:
* @trans: a #GstRTSPStreamTransport
rtsp: only create transport when needed Only create the StreamTransport when configured.
2012-10-27 22:23:57 +00:00
* @tr: (transfer full): a client #GstRTSPTransport
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
*
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
* Set @tr as the client transport. This function takes ownership of the
* passed @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: 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
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_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)
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: 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: 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
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
rtsp: only create transport when needed Only create the StreamTransport when configured.
2012-10-27 22:23:57 +00:00
g_return_if_fail (tr != NULL);
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: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv = trans->priv;
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
/* keep track of the transports in the stream. */
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
if (priv->transport)
gst_rtsp_transport_free (priv->transport);
priv->transport = tr;
}
/**
* gst_rtsp_stream_transport_get_transport:
* @trans: a #GstRTSPStreamTransport
*
* Get the transport configured in @trans.
*
All around: add annotations and API guards
2018-02-09 23:07:25 +00:00
* Returns: (transfer none) (nullable): the transport configured in @trans. It remains
gobject-introspection: Add annotations to support language bindings In addition a few cosmetic changes: * Adjust the order of arguments * Fix typo: occured -> occurred * Fix indentation after Return:-clauses Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-23 20:24:48 +00:00
* valid for as long as @trans is valid.
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)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
return trans->priv->transport;
}
stream-transport: add method to get/set url
2013-11-28 16:35:45 +00:00
/**
* gst_rtsp_stream_transport_set_url:
* @trans: a #GstRTSPStreamTransport
All around: add annotations and API guards
2018-02-09 23:07:25 +00:00
* @url: (transfer none) (nullable): a client #GstRTSPUrl
stream-transport: add method to get/set url
2013-11-28 16:35:45 +00:00
*
* Set @url as the client url.
*/
void
gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans,
const GstRTSPUrl * url)
{
GstRTSPStreamTransportPrivate *priv;
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
priv = trans->priv;
/* keep track of the transports in the stream. */
if (priv->url)
gst_rtsp_url_free (priv->url);
priv->url = (url ? gst_rtsp_url_copy (url) : NULL);
}
/**
* gst_rtsp_stream_transport_get_url:
* @trans: a #GstRTSPStreamTransport
*
* Get the url configured in @trans.
*
All around: add annotations and API guards
2018-02-09 23:07:25 +00:00
* Returns: (transfer none) (nullable): the url configured in @trans.
* It remains valid for as long as @trans is valid.
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)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
return trans->priv->url;
}
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
/**
* gst_rtsp_stream_transport_get_rtpinfo:
* @trans: a #GstRTSPStreamTransport
session-media: calculate start-time
2013-12-26 15:28:59 +00:00
* @start_time: a star 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
*
session: improve RTP-Info Ignore streams that can't generate RTP-Info instead of failing. Don't return the empty string when all streams are unconfigured but return NULL so that we don't generate and empty RTP-Info header. Improve docs a little.
2014-02-07 15:39:49 +00:00
* Get the RTP-Info string for @trans and @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
*
introspection: add (nullable) annotations to return values https://bugzilla.gnome.org/show_bug.cgi?id=730952
2014-06-11 23:38:36 +00:00
* Returns: (transfer full) (nullable): the RTPInfo string for @trans
* and @start_time or %NULL when the RTP-Info could not be
* determined. g_free() after usage.
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
*/
gchar *
session-media: calculate start-time
2013-12-26 15:28:59 +00:00
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
{
GstRTSPStreamTransportPrivate *priv;
gchar *url_str;
GString *rtpinfo;
stream: return clock-rate from get_rtpinfo And use it to correct the rtptime to the requested start-time. See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 16:02:50 +00:00
guint rtptime, seq, clock_rate;
GstClockTime running_time = GST_CLOCK_TIME_NONE;
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
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), NULL);
priv = trans->priv;
Get payloader stats only for the sending streams Get/set payloader properties only for streams that actually contain a payloader element. https://bugzilla.gnome.org/show_bug.cgi?id=796523
2018-06-05 06:44:44 +00:00
if (!gst_rtsp_stream_is_sender (priv->stream))
return NULL;
stream: return clock-rate from get_rtpinfo And use it to correct the rtptime to the requested start-time. See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 16:02:50 +00:00
if (!gst_rtsp_stream_get_rtpinfo (priv->stream, &rtptime, &seq, &clock_rate,
&running_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
return NULL;
stream: return clock-rate from get_rtpinfo And use it to correct the rtptime to the requested start-time. See https://bugzilla.gnome.org/show_bug.cgi?id=712198
2013-12-26 16:02:50 +00:00
GST_DEBUG ("RTP time %u, seq %u, rate %u, running-time %" GST_TIME_FORMAT,
rtptime, seq, clock_rate, GST_TIME_ARGS (running_time));
if (GST_CLOCK_TIME_IS_VALID (running_time)
&& GST_CLOCK_TIME_IS_VALID (start_time)) {
if (running_time > start_time) {
rtptime -=
gst_util_uint64_scale_int (running_time - start_time, clock_rate,
GST_SECOND);
} else {
rtptime +=
gst_util_uint64_scale_int (start_time - running_time, clock_rate,
GST_SECOND);
}
}
GST_DEBUG ("RTP time %u, for start-time %" GST_TIME_FORMAT,
rtptime, GST_TIME_ARGS (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
rtpinfo = g_string_new ("");
url_str = gst_rtsp_url_get_request_uri (trans->priv->url);
g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
url_str, seq, rtptime);
g_free (url_str);
return g_string_free (rtpinfo, FALSE);
}
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
/**
* gst_rtsp_stream_transport_set_active:
* @trans: a #GstRTSPStreamTransport
* @active: new state of @trans
*
* Activate or deactivate datatransfer configured in @trans.
*
* Returns: %TRUE when the state was changed.
*/
gboolean
gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans,
gboolean active)
{
GstRTSPStreamTransportPrivate *priv;
gboolean res;
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
priv = trans->priv;
if (active)
res = gst_rtsp_stream_add_transport (priv->stream, trans);
else
res = gst_rtsp_stream_remove_transport (priv->stream, trans);
return res;
}
/**
* gst_rtsp_stream_transport_set_timed_out:
* @trans: a #GstRTSPStreamTransport
* @timedout: timed out value
*
* Set the timed out state of @trans to @timedout
*/
void
gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans,
gboolean timedout)
{
g_return_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans));
trans->priv->timed_out = timedout;
}
/**
* gst_rtsp_stream_transport_is_timed_out:
* @trans: a #GstRTSPStreamTransport
*
* Check if @trans is timed out.
*
* Returns: %TRUE if @trans timed out.
*/
gboolean
gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
{
g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
return trans->priv->timed_out;
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
}
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
/**
* gst_rtsp_stream_transport_send_rtp:
* @trans: a #GstRTSPStreamTransport
gobject-introspection: Add annotations to support language bindings In addition a few cosmetic changes: * Adjust the order of arguments * Fix typo: occured -> occurred * Fix indentation after Return:-clauses Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-23 20:24:48 +00:00
* @buffer: (transfer none): a #GstBuffer
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
*
* Send @buffer to the installed RTP callback for @trans.
*
* Returns: %TRUE on success
*/
gboolean
gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
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;
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 res = FALSE;
All around: add annotations and API guards
2018-02-09 23:07:25 +00:00
g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv = trans->priv;
if (priv->send_rtp)
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
res =
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv->send_rtp (buffer, priv->transport->interleaved.min,
priv->user_data);
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
rtsp-*: Treat sending packets to clients as keepalive As long as gst-rtsp-server can successfully send RTP/RTCP data to clients then the client must be reading. This change makes the server timeout the connection if the client stops reading. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-19 16:28:50 +00:00
if (res)
gst_rtsp_stream_transport_keep_alive (trans);
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
return res;
}
/**
* gst_rtsp_stream_transport_send_rtcp:
* @trans: a #GstRTSPStreamTransport
gobject-introspection: Add annotations to support language bindings In addition a few cosmetic changes: * Adjust the order of arguments * Fix typo: occured -> occurred * Fix indentation after Return:-clauses Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726941
2014-03-23 20:24:48 +00:00
* @buffer: (transfer none): a #GstBuffer
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
*
* Send @buffer to the installed RTCP callback for @trans.
*
* Returns: %TRUE on success
*/
gboolean
gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans,
GstBuffer * buffer)
{
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;
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 res = FALSE;
All around: add annotations and API guards
2018-02-09 23:07:25 +00:00
g_return_val_if_fail (GST_IS_BUFFER (buffer), FALSE);
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv = trans->priv;
if (priv->send_rtcp)
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
res =
rtsp: make object details private Make all object details private Add methods to access private bits
2012-11-29 10:11:05 +00:00
priv->send_rtcp (buffer, priv->transport->interleaved.max,
priv->user_data);
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
rtsp-*: Treat sending packets to clients as keepalive As long as gst-rtsp-server can successfully send RTP/RTCP data to clients then the client must be reading. This change makes the server timeout the connection if the client stops reading. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=736647
2014-09-19 16:28:50 +00:00
if (res)
gst_rtsp_stream_transport_keep_alive (trans);
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
return res;
}
stream-transport: add keep-alive method
2012-11-12 16:11:18 +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_stream_transport_send_rtp_list:
* @trans: a #GstRTSPStreamTransport
* @buffer_list: (transfer none): a #GstBufferList
*
* Send @buffer_list to the installed RTP callback for @trans.
*
* Returns: %TRUE on success
*
* Since: 1.16
*/
gboolean
gst_rtsp_stream_transport_send_rtp_list (GstRTSPStreamTransport * trans,
GstBufferList * buffer_list)
{
GstRTSPStreamTransportPrivate *priv;
gboolean res = FALSE;
g_return_val_if_fail (GST_IS_BUFFER_LIST (buffer_list), FALSE);
priv = trans->priv;
if (priv->send_rtp_list) {
res =
priv->send_rtp_list (buffer_list, priv->transport->interleaved.min,
priv->list_user_data);
} else if (priv->send_rtp) {
guint n = gst_buffer_list_length (buffer_list), i;
for (i = 0; i < n; i++) {
GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
res =
priv->send_rtp (buffer, priv->transport->interleaved.min,
priv->user_data);
if (!res)
break;
}
}
if (res)
gst_rtsp_stream_transport_keep_alive (trans);
return res;
}
/**
* gst_rtsp_stream_transport_send_rtcp_list:
* @trans: a #GstRTSPStreamTransport
* @buffer_list: (transfer none): a #GstBuffer
*
* Send @buffer_list to the installed RTCP callback for @trans.
*
* Returns: %TRUE on success
*
* Since: 1.16
*/
gboolean
gst_rtsp_stream_transport_send_rtcp_list (GstRTSPStreamTransport * trans,
GstBufferList * buffer_list)
{
GstRTSPStreamTransportPrivate *priv;
gboolean res = FALSE;
g_return_val_if_fail (GST_IS_BUFFER_LIST (buffer_list), FALSE);
priv = trans->priv;
if (priv->send_rtcp_list) {
res =
priv->send_rtcp_list (buffer_list, priv->transport->interleaved.max,
priv->list_user_data);
} else if (priv->send_rtcp) {
guint n = gst_buffer_list_length (buffer_list), i;
for (i = 0; i < n; i++) {
GstBuffer *buffer = gst_buffer_list_get (buffer_list, i);
res =
priv->send_rtcp (buffer, priv->transport->interleaved.max,
priv->user_data);
if (!res)
break;
}
}
if (res)
gst_rtsp_stream_transport_keep_alive (trans);
return res;
}
stream-transport: add keep-alive method
2012-11-12 16:11:18 +00:00
/**
* gst_rtsp_stream_transport_keep_alive:
* @trans: a #GstRTSPStreamTransport
*
* Signal the installed keep_alive callback for @trans.
*/
void
gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
{
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;
priv = trans->priv;
if (priv->keep_alive)
priv->keep_alive (priv->ka_user_data);
stream-transport: add keep-alive method
2012-11-12 16:11:18 +00:00
}
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
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_stream_transport_message_sent:
* @trans: a #GstRTSPStreamTransport
*
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
* Signal the installed message_sent / message_sent_full callback for @trans.
rtsp-server: Add various missing Since: 1.16 markers
2019-04-23 11:38:05 +00:00
*
* Since: 1.16
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
*/
void
gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport * trans)
{
GstRTSPStreamTransportPrivate *priv;
priv = trans->priv;
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
if (priv->message_sent_full)
priv->message_sent_full (trans, priv->msf_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
if (priv->message_sent)
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
priv->message_sent (priv->ms_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
}
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
/**
* gst_rtsp_stream_transport_recv_data:
* @trans: a #GstRTSPStreamTransport
* @channel: a channel
* @buffer: (transfer full): a #GstBuffer
*
* Receive @buffer on @channel @trans.
*
* Returns: a #GstFlowReturn. Returns GST_FLOW_NOT_LINKED when @channel is not
* configured in the transport of @trans.
*/
GstFlowReturn
gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport * trans,
guint channel, GstBuffer * buffer)
{
GstRTSPStreamTransportPrivate *priv;
const GstRTSPTransport *tr;
GstFlowReturn res;
All around: add annotations and API guards
2018-02-09 23:07:25 +00:00
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
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
priv = trans->priv;
tr = priv->transport;
if (tr->interleaved.min == channel) {
res = gst_rtsp_stream_recv_rtp (priv->stream, buffer);
} else if (tr->interleaved.max == channel) {
res = gst_rtsp_stream_recv_rtcp (priv->stream, buffer);
} else {
res = GST_FLOW_NOT_LINKED;
}
return res;
}
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
static GstClockTime
get_backlog_item_timestamp (BackLogItem * item)
{
GstClockTime ret = GST_CLOCK_TIME_NONE;
if (item->buffer) {
ret = GST_BUFFER_DTS_OR_PTS (item->buffer);
} else if (item->buffer_list) {
g_assert (gst_buffer_list_length (item->buffer_list) > 0);
ret = GST_BUFFER_DTS_OR_PTS (gst_buffer_list_get (item->buffer_list, 0));
}
return ret;
}
static GstClockTime
get_first_backlog_timestamp (GstRTSPStreamTransport * trans)
{
GstRTSPStreamTransportPrivate *priv = trans->priv;
GstClockTime ret = GST_CLOCK_TIME_NONE;
guint i, l;
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
l = gst_vec_deque_get_length (priv->items);
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
for (i = 0; i < l; i++) {
BackLogItem *item = (BackLogItem *)
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
gst_vec_deque_peek_nth_struct (priv->items, i);
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
if (item->is_rtp) {
ret = get_backlog_item_timestamp (item);
break;
}
}
return ret;
}
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
/* Not MT-safe, caller should ensure consistent locking (see
* gst_rtsp_stream_transport_lock_backlog()). Ownership
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
* of @buffer and @buffer_list is transfered to the transport */
gboolean
gst_rtsp_stream_transport_backlog_push (GstRTSPStreamTransport * trans,
GstBuffer * buffer, GstBufferList * buffer_list, gboolean is_rtp)
{
gboolean ret = TRUE;
BackLogItem item = { 0, };
GstClockTime item_timestamp;
GstRTSPStreamTransportPrivate *priv;
priv = trans->priv;
if (buffer)
item.buffer = buffer;
if (buffer_list)
item.buffer_list = buffer_list;
item.is_rtp = is_rtp;
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
gst_vec_deque_push_tail_struct (priv->items, &item);
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
item_timestamp = get_backlog_item_timestamp (&item);
if (is_rtp && priv->first_rtp_timestamp != GST_CLOCK_TIME_NONE) {
GstClockTimeDiff queue_duration;
g_assert (GST_CLOCK_TIME_IS_VALID (item_timestamp));
queue_duration = GST_CLOCK_DIFF (priv->first_rtp_timestamp, item_timestamp);
g_assert (queue_duration >= 0);
if (queue_duration > MAX_BACKLOG_DURATION &&
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
gst_vec_deque_get_length (priv->items) > MAX_BACKLOG_SIZE) {
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
ret = FALSE;
}
} else if (is_rtp) {
priv->first_rtp_timestamp = item_timestamp;
}
return ret;
}
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
/* Not MT-safe, caller should ensure consistent locking (see
* gst_rtsp_stream_transport_lock_backlog()). Ownership
rtsp-stream: clear backlog when removing transport This ensures we don't end up calling any of transports' callbacks with a potentially unreffed user_data (in practice, a client that may have been removed)
2020-02-14 13:59:25 +00:00
* of @buffer and @buffer_list is transfered back to the caller,
* if either of those is NULL the underlying object is unreffed */
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
gboolean
gst_rtsp_stream_transport_backlog_pop (GstRTSPStreamTransport * trans,
GstBuffer ** buffer, GstBufferList ** buffer_list, gboolean * is_rtp)
{
BackLogItem *item;
GstRTSPStreamTransportPrivate *priv;
g_return_val_if_fail (!gst_rtsp_stream_transport_backlog_is_empty (trans),
FALSE);
priv = trans->priv;
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
item = (BackLogItem *) gst_vec_deque_pop_head_struct (priv->items);
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
priv->first_rtp_timestamp = get_first_backlog_timestamp (trans);
rtsp-stream: clear backlog when removing transport This ensures we don't end up calling any of transports' callbacks with a potentially unreffed user_data (in practice, a client that may have been removed)
2020-02-14 13:59:25 +00:00
if (buffer)
*buffer = item->buffer;
else if (item->buffer)
gst_buffer_unref (item->buffer);
if (buffer_list)
*buffer_list = item->buffer_list;
else if (item->buffer_list)
gst_buffer_list_unref (item->buffer_list);
if (is_rtp)
*is_rtp = item->is_rtp;
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
return TRUE;
}
gst-rtsp-server: Fix pushing backlog to client Check back pressure of a stream transport before popping buffer from its backlog. If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client. Fixes:#1298 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
2022-08-24 11:50:19 +00:00
/* Not MT-safe, caller should ensure consistent locking.
* See gst_rtsp_stream_transport_lock_backlog() */
gboolean
gst_rtsp_stream_transport_backlog_peek_is_rtp (GstRTSPStreamTransport * trans)
{
BackLogItem *item;
GstRTSPStreamTransportPrivate *priv;
g_return_val_if_fail (!gst_rtsp_stream_transport_backlog_is_empty (trans),
FALSE);
priv = trans->priv;
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
item = (BackLogItem *) gst_vec_deque_peek_head_struct (priv->items);
gst-rtsp-server: Fix pushing backlog to client Check back pressure of a stream transport before popping buffer from its backlog. If the stream transport is not experiencing back pressure, the buffer can be popped from backlog and pushed to client. Fixes:#1298 Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2936>
2022-08-24 11:50:19 +00:00
return item->is_rtp;
}
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
/* Not MT-safe, caller should ensure consistent locking.
* See gst_rtsp_stream_transport_lock_backlog() */
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
gboolean
gst_rtsp_stream_transport_backlog_is_empty (GstRTSPStreamTransport * trans)
{
gst: Move GstQueueArray as GstVecDeque to core And change lengths and indices from guint to gsize for a more correct type. Also deprecate GstQueueArray and implement it in terms of GstVecDeque. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-04-18 14:07:25 +00:00
return gst_vec_deque_is_empty (trans->priv->items);
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
}
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
rtsp-stream: clear backlog when removing transport This ensures we don't end up calling any of transports' callbacks with a potentially unreffed user_data (in practice, a client that may have been removed)
2020-02-14 13:59:25 +00:00
/* Not MT-safe, caller should ensure consistent locking.
* See gst_rtsp_stream_transport_lock_backlog() */
void
gst_rtsp_stream_transport_clear_backlog (GstRTSPStreamTransport * trans)
{
while (!gst_rtsp_stream_transport_backlog_is_empty (trans)) {
gst_rtsp_stream_transport_backlog_pop (trans, NULL, NULL, NULL);
}
}
rtsp-stream: properly protect TCP backlog access Fixes #97 We cannot hold stream->lock while pushing data, but need to consistently check the state of the backlog both from the send_tcp_message function and the on_message_sent function, which may or may not be called from the same thread. This commit introduces internal API to allow for potentially recursive locking of transport streams, addressing a race condition where the RTSP stream could push items out of order when popping them from the backlog.
2020-02-05 19:28:19 +00:00
/* Internal API, protects access to the TCP backlog. Safe to
* call recursively */
void
gst_rtsp_stream_transport_lock_backlog (GstRTSPStreamTransport * trans)
{
g_rec_mutex_lock (&trans->priv->backlog_lock);
}
/* See gst_rtsp_stream_transport_lock_backlog() */
void
gst_rtsp_stream_transport_unlock_backlog (GstRTSPStreamTransport * trans)
{
g_rec_mutex_unlock (&trans->priv->backlog_lock);
}
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