mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 21:51:09 +00:00
52c676546d
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
88 lines
3 KiB
C
88 lines
3 KiB
C
/* GStreamer
|
|
* Copyright (C) 2017 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 __GST_WEBRTC_RTP_SENDER_H__
|
|
#define __GST_WEBRTC_RTP_SENDER_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/webrtc/webrtc_fwd.h>
|
|
#include <gst/webrtc/dtlstransport.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
GST_WEBRTC_API
|
|
GType gst_webrtc_rtp_sender_get_type(void);
|
|
#define GST_TYPE_WEBRTC_RTP_SENDER (gst_webrtc_rtp_sender_get_type())
|
|
#define GST_WEBRTC_RTP_SENDER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSender))
|
|
#define GST_IS_WEBRTC_RTP_SENDER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_RTP_SENDER))
|
|
#define GST_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
|
|
#define GST_IS_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_SENDER))
|
|
#define GST_WEBRTC_RTP_SENDER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass))
|
|
|
|
/**
|
|
* GstWebRTCRTPSender:
|
|
* @transport: The transport for RTP packets
|
|
* @send_encodings: Unused
|
|
* @priority: The priority of the stream (Since: 1.20)
|
|
*
|
|
* An object to track the sending aspect of the stream
|
|
*
|
|
* Mostly matches the WebRTC RTCRtpSender interface.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
/**
|
|
* GstWebRTCRTPSender.priority:
|
|
*
|
|
* The priority of the stream
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
struct _GstWebRTCRTPSender
|
|
{
|
|
GstObject parent;
|
|
|
|
/* The MediStreamTrack is represented by the stream and is output into @transport as necessary */
|
|
GstWebRTCDTLSTransport *transport;
|
|
|
|
GArray *send_encodings;
|
|
GstWebRTCPriorityType priority;
|
|
|
|
gpointer _padding[GST_PADDING];
|
|
};
|
|
|
|
struct _GstWebRTCRTPSenderClass
|
|
{
|
|
GstObjectClass parent_class;
|
|
|
|
gpointer _padding[GST_PADDING];
|
|
};
|
|
|
|
GST_WEBRTC_API
|
|
GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void);
|
|
|
|
GST_WEBRTC_API
|
|
void gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender *sender,
|
|
GstWebRTCPriorityType priority);
|
|
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPSender, gst_object_unref)
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_WEBRTC_RTP_SENDER_H__ */
|