#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/dtlstransport.h>
+#include "webrtc-priv.h"
+
G_BEGIN_DECLS
GST_WEBRTC_API
#define GST_IS_WEBRTC_RTP_RECEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_RECEIVER))
#define GST_WEBRTC_RTP_RECEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_RECEIVER,GstWebRTCRTPReceiverClass))
-/**
- * GstWebRTCRTPReceiver:
- * @transport: The transport for RTP packets
- *
- * An object to track the receiving aspect of the stream
- *
- * Mostly matches the WebRTC RTCRtpReceiver interface.
- *
- * Since: 1.16
- */
-struct _GstWebRTCRTPReceiver
-{
- GstObject parent;
-
- /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */
- GstWebRTCDTLSTransport *transport;
-
- gpointer _padding[GST_PADDING];
-};
-
-struct _GstWebRTCRTPReceiverClass
-{
- GstObjectClass parent_class;
-
- gpointer _padding[GST_PADDING];
-};
-
-GST_WEBRTC_API
-GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
-
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPReceiver, gst_object_unref)
G_END_DECLS
GST_WEBRTC_API
GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void);
+/**
+ * GstWebRTCRTPReceiver:
+ * @transport: The transport for RTP packets
+ *
+ * An object to track the receiving aspect of the stream
+ *
+ * Mostly matches the WebRTC RTCRtpReceiver interface.
+ *
+ * Since: 1.16
+ */
+struct _GstWebRTCRTPReceiver
+{
+ GstObject parent;
+
+ /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */
+ GstWebRTCDTLSTransport *transport;
+
+ gpointer _padding[GST_PADDING];
+};
+
+struct _GstWebRTCRTPReceiverClass
+{
+ GstObjectClass parent_class;
+
+ gpointer _padding[GST_PADDING];
+};
+
+GST_WEBRTC_API
+GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
+
+
G_END_DECLS
#endif /* __GST_WEBRTC_PRIV_H__ */