mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 15:51:11 +00:00
d708f9736b
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
214 lines
8.8 KiB
C
214 lines
8.8 KiB
C
/* 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
|
|
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/rtsp/gstrtsprange.h>
|
|
#include <gst/rtsp/gstrtspurl.h>
|
|
|
|
#ifndef __GST_RTSP_STREAM_TRANSPORT_H__
|
|
#define __GST_RTSP_STREAM_TRANSPORT_H__
|
|
|
|
#include "rtsp-server-prelude.h"
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
/* types for the media */
|
|
#define GST_TYPE_RTSP_STREAM_TRANSPORT (gst_rtsp_stream_transport_get_type ())
|
|
#define GST_IS_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT))
|
|
#define GST_IS_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT))
|
|
#define GST_RTSP_STREAM_TRANSPORT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
|
|
#define GST_RTSP_STREAM_TRANSPORT(obj) (G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransport))
|
|
#define GST_RTSP_STREAM_TRANSPORT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_RTSP_STREAM_TRANSPORT, GstRTSPStreamTransportClass))
|
|
#define GST_RTSP_STREAM_TRANSPORT_CAST(obj) ((GstRTSPStreamTransport*)(obj))
|
|
#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
|
|
|
|
typedef struct _GstRTSPStreamTransport GstRTSPStreamTransport;
|
|
typedef struct _GstRTSPStreamTransportClass GstRTSPStreamTransportClass;
|
|
typedef struct _GstRTSPStreamTransportPrivate GstRTSPStreamTransportPrivate;
|
|
|
|
#include "rtsp-stream.h"
|
|
|
|
/**
|
|
* GstRTSPSendFunc:
|
|
* @buffer: a #GstBuffer
|
|
* @channel: a channel
|
|
* @user_data: user data
|
|
*
|
|
* Function registered with gst_rtsp_stream_transport_set_callbacks() and
|
|
* called when @buffer must be sent on @channel.
|
|
*
|
|
* Returns: %TRUE on success
|
|
*/
|
|
typedef gboolean (*GstRTSPSendFunc) (GstBuffer *buffer, guint8 channel, gpointer user_data);
|
|
|
|
/**
|
|
* GstRTSPSendListFunc:
|
|
* @buffer_list: a #GstBufferList
|
|
* @channel: a channel
|
|
* @user_data: user data
|
|
*
|
|
* Function registered with gst_rtsp_stream_transport_set_callbacks() and
|
|
* called when @buffer_list must be sent on @channel.
|
|
*
|
|
* Returns: %TRUE on success
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
typedef gboolean (*GstRTSPSendListFunc) (GstBufferList *buffer_list, guint8 channel, gpointer user_data);
|
|
|
|
/**
|
|
* GstRTSPKeepAliveFunc:
|
|
* @user_data: user data
|
|
*
|
|
* Function registered with gst_rtsp_stream_transport_set_keepalive() and called
|
|
* when the stream is active.
|
|
*/
|
|
typedef void (*GstRTSPKeepAliveFunc) (gpointer user_data);
|
|
|
|
/**
|
|
* GstRTSPMessageSentFunc:
|
|
* @user_data: user data
|
|
*
|
|
* Function registered with gst_rtsp_stream_transport_set_message_sent()
|
|
* and called when a message has been sent on the transport.
|
|
*/
|
|
typedef void (*GstRTSPMessageSentFunc) (gpointer user_data);
|
|
|
|
/**
|
|
* GstRTSPStreamTransport:
|
|
* @parent: parent instance
|
|
*
|
|
* A Transport description for a stream
|
|
*/
|
|
struct _GstRTSPStreamTransport {
|
|
GObject parent;
|
|
|
|
/*< private >*/
|
|
GstRTSPStreamTransportPrivate *priv;
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
struct _GstRTSPStreamTransportClass {
|
|
GObjectClass parent_class;
|
|
|
|
/*< private >*/
|
|
gpointer _gst_reserved[GST_PADDING];
|
|
};
|
|
|
|
GST_RTSP_SERVER_API
|
|
GType gst_rtsp_stream_transport_get_type (void);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream *stream,
|
|
GstRTSPTransport *tr);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstRTSPStream * gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport *trans);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport *trans,
|
|
GstRTSPTransport * tr);
|
|
|
|
GST_RTSP_SERVER_API
|
|
const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport *trans);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport *trans,
|
|
const GstRTSPUrl * url);
|
|
|
|
GST_RTSP_SERVER_API
|
|
const GstRTSPUrl * gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport *trans);
|
|
|
|
|
|
GST_RTSP_SERVER_API
|
|
gchar * gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport *trans,
|
|
GstClockTime start_time);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport *trans,
|
|
GstRTSPSendFunc send_rtp,
|
|
GstRTSPSendFunc send_rtcp,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_set_list_callbacks (GstRTSPStreamTransport *trans,
|
|
GstRTSPSendListFunc send_rtp_list,
|
|
GstRTSPSendListFunc send_rtcp_list,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport *trans,
|
|
GstRTSPKeepAliveFunc keep_alive,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport *trans,
|
|
GstRTSPMessageSentFunc message_sent,
|
|
gpointer user_data,
|
|
GDestroyNotify notify);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport *trans);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport *trans);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport *trans,
|
|
gboolean active);
|
|
|
|
GST_RTSP_SERVER_API
|
|
void gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport *trans,
|
|
gboolean timedout);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport *trans);
|
|
|
|
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport *trans,
|
|
GstBuffer *buffer);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport *trans,
|
|
GstBuffer *buffer);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_transport_send_rtp_list (GstRTSPStreamTransport *trans,
|
|
GstBufferList *buffer_list);
|
|
|
|
GST_RTSP_SERVER_API
|
|
gboolean gst_rtsp_stream_transport_send_rtcp_list(GstRTSPStreamTransport *trans,
|
|
GstBufferList *buffer_list);
|
|
|
|
GST_RTSP_SERVER_API
|
|
GstFlowReturn gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport *trans,
|
|
guint channel, GstBuffer *buffer);
|
|
|
|
#ifdef G_DEFINE_AUTOPTR_CLEANUP_FUNC
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstRTSPStreamTransport, gst_object_unref)
|
|
#endif
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_RTSP_STREAM_TRANSPORT_H__ */
|