gstreamer/gst-libs/gst/webrtc/rtpsender.h

/* 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
 * @rtcp_transport: The transport for RTCP packets without rtcp-mux
 * @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/@rtcp_transport as necessary */
  GstWebRTCDTLSTransport            *transport;
  GstWebRTCDTLSTransport            *rtcp_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_transport         (GstWebRTCRTPSender * sender,
                                                                         GstWebRTCDTLSTransport * transport);
GST_WEBRTC_API
void                        gst_webrtc_rtp_sender_set_rtcp_transport    (GstWebRTCRTPSender * sender,
                                                                         GstWebRTCDTLSTransport * transport);
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__ */