gstreamer/ext/webrtc/webrtcdatachannel.h
Johan Sternerup 607ef6db60 webrtc: Split sctptransport into lib and implementation parts
GstWebRTCSCTPTransport is now made into into an abstract base class
that only contains property specifications matching the
RTCSctpTransport interface of the W3C WebRTC specification, see
https://w3c.github.io/webrtc-pc/#rtcsctptransport-interface. This
class is put into the WebRTC library to expose it for applications and
to allow for generation of bindings for non-dynamic languages using
GObject introspection.

The actual implementation is moved to the subclass WebRTCSCTPTransport
located in the WebRTC plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2214>
2021-08-25 13:20:22 +00:00

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2.8 KiB
C

/* GStreamer
* Copyright (C) 2018 Matthew Waters <matthew@centricular.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.
*/
#ifndef __WEBRTC_DATA_CHANNEL_H__
#define __WEBRTC_DATA_CHANNEL_H__
#include <gst/gst.h>
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/dtlstransport.h>
#include <gst/webrtc/datachannel.h>
#include "webrtcsctptransport.h"
#include "gst/webrtc/webrtc-priv.h"
G_BEGIN_DECLS
GType webrtc_data_channel_get_type(void);
#define WEBRTC_TYPE_DATA_CHANNEL (webrtc_data_channel_get_type())
#define WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),WEBRTC_TYPE_DATA_CHANNEL,WebRTCDataChannel))
#define WEBRTC_IS_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),WEBRTC_TYPE_DATA_CHANNEL))
#define WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,WEBRTC_TYPE_DATA_CHANNEL,WebRTCDataChannelClass))
#define WEBRTC_IS_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,WEBRTC_TYPE_DATA_CHANNEL))
#define WEBRTC_DATA_CHANNEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,WEBRTC_TYPE_DATA_CHANNEL,WebRTCDataChannelClass))
typedef struct _WebRTCDataChannel WebRTCDataChannel;
typedef struct _WebRTCDataChannelClass WebRTCDataChannelClass;
struct _WebRTCDataChannel
{
GstWebRTCDataChannel parent;
WebRTCSCTPTransport *sctp_transport;
GstElement *appsrc;
GstElement *appsink;
GstWebRTCBin *webrtcbin;
gboolean opened;
gulong src_probe;
GError *stored_error;
gboolean peer_closed;
gpointer _padding[GST_PADDING];
};
struct _WebRTCDataChannelClass
{
GstWebRTCDataChannelClass parent_class;
gpointer _padding[GST_PADDING];
};
void webrtc_data_channel_start_negotiation (WebRTCDataChannel *channel);
G_GNUC_INTERNAL
void webrtc_data_channel_link_to_sctp (WebRTCDataChannel *channel,
WebRTCSCTPTransport *sctp_transport);
G_END_DECLS
#endif /* __WEBRTC_DATA_CHANNEL_H__ */