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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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12169f1e84
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
184 lines
7.5 KiB
C
184 lines
7.5 KiB
C
/* GStreamer
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* Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <gst/gst.h>
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#include <gst/rtsp/gstrtsprange.h>
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#include <gst/rtsp/gstrtspurl.h>
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#ifndef __GST_RTSP_STREAM_TRANSPORT_H__
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#define __GST_RTSP_STREAM_TRANSPORT_H__
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#include "rtsp-server-prelude.h"
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G_BEGIN_DECLS
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/* types for the media */
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#define GST_TYPE_RTSP_STREAM_TRANSPORT (gst_rtsp_stream_transport_get_type ())
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#define GST_IS_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT))
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#define GST_IS_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT))
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#define GST_RTSP_STREAM_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
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#define GST_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransport))
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#define GST_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
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#define GST_RTSP_STREAM_TRANSPORT_CAST(obj) ((GstRTSPStreamTransport*)(obj))
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#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
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typedef struct _GstRTSPStreamTransport GstRTSPStreamTransport;
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typedef struct _GstRTSPStreamTransportClass GstRTSPStreamTransportClass;
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typedef struct _GstRTSPStreamTransportPrivate GstRTSPStreamTransportPrivate;
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#include "rtsp-stream.h"
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/**
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* GstRTSPSendFunc:
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* @buffer: a #GstBuffer
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* @channel: a channel
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* @user_data: user data
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*
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* Function registered with gst_rtsp_stream_transport_set_callbacks() and
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* called when @buffer must be sent on @channel.
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*
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* Returns: %TRUE on success
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*/
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typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
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/**
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* GstRTSPKeepAliveFunc:
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* @user_data: user data
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*
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* Function registered with gst_rtsp_stream_transport_set_keepalive() and called
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* when the stream is active.
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*/
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typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
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/**
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* GstRTSPMessageSentFunc:
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* @user_data: user data
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*
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* Function registered with gst_rtsp_stream_transport_set_message_sent()
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* and called when a message has been sent on the transport.
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*/
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typedef void (*GstRTSPMessageSentFunc) (gpointer user_data);
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/**
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* GstRTSPStreamTransport:
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* @parent: parent instance
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*
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* A Transport description for a stream
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*/
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struct _GstRTSPStreamTransport {
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GObject parent;
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/*< private >*/
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GstRTSPStreamTransportPrivate *priv;
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gpointer _gst_reserved[GST_PADDING];
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};
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struct _GstRTSPStreamTransportClass {
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GObjectClass parent_class;
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/*< private >*/
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gpointer _gst_reserved[GST_PADDING];
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};
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GST_RTSP_SERVER_API
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GType gst_rtsp_stream_transport_get_type (void);
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GST_RTSP_SERVER_API
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GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream,
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GstRTSPTransport *tr);
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GST_RTSP_SERVER_API
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GstRTSPStream * gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport *trans);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
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GstRTSPTransport * tr);
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GST_RTSP_SERVER_API
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const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport *trans);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport *trans,
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const GstRTSPUrl * url);
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GST_RTSP_SERVER_API
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const GstRTSPUrl * gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport *trans);
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GST_RTSP_SERVER_API
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gchar * gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport *trans,
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GstClockTime start_time);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
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GstRTSPSendFunc send_rtp,
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GstRTSPSendFunc send_rtcp,
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gpointer user_data,
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GDestroyNotify notify);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport *trans,
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GstRTSPKeepAliveFunc keep_alive,
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gpointer user_data,
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GDestroyNotify notify);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport *trans,
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GstRTSPMessageSentFunc message_sent,
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gpointer user_data,
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GDestroyNotify notify);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport *trans);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport *trans);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport *trans,
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gboolean active);
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GST_RTSP_SERVER_API
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void gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport *trans,
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gboolean timedout);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport *trans);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
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GstBuffer *buffer);
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GST_RTSP_SERVER_API
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gboolean gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
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GstBuffer *buffer);
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GST_RTSP_SERVER_API
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GstFlowReturn gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport *trans,
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guint channel, GstBuffer *buffer);
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#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPStreamTransport, gst_object_unref)
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#endif
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G_END_DECLS
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#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */
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