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/* 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))
struct _GstWebRTCRTPSender
{
GstObject parent;
/* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */
GstWebRTCDTLSTransport *transport;
GstWebRTCDTLSTransport *rtcp_transport;
GArray *send_encodings;
gpointer _padding[GST_PADDING];
};
struct _GstWebRTCRTPSenderClass
{
GstObjectClass parent_class;
gpointer _padding[GST_PADDING];
};
GST_WEBRTC_API
GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (GArray * send_encodings);
GST_WEBRTC_API
GstStructure * gst_webrtc_rtp_sender_get_parameters (GstWebRTCRTPSender * sender, gchar * kind);
/* FIXME: promise? */
GST_WEBRTC_API
gboolean gst_webrtc_rtp_sender_set_parameters (GstWebRTCRTPSender * sender,
GstStructure * parameters);
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport);
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport);
G_END_DECLS
#endif /* __GST_WEBRTC_RTP_SENDER_H__ */
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