/* GStreamer * Copyright (C) 2018 Matthew Waters * * 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_DATA_CHANNEL_H__ #define __GST_WEBRTC_DATA_CHANNEL_H__ #include #include #include #include "sctptransport.h" G_BEGIN_DECLS GST_WEBRTC_API GType gst_webrtc_data_channel_get_type(void); #define GST_TYPE_WEBRTC_DATA_CHANNEL (gst_webrtc_data_channel_get_type()) #define GST_WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannel)) #define GST_IS_WEBRTC_DATA_CHANNEL(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_DATA_CHANNEL)) #define GST_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass)) #define GST_IS_WEBRTC_DATA_CHANNEL_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_DATA_CHANNEL)) #define GST_WEBRTC_DATA_CHANNEL_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_DATA_CHANNEL,GstWebRTCDataChannelClass)) typedef struct _GstWebRTCDataChannel GstWebRTCDataChannel; typedef struct _GstWebRTCDataChannelClass GstWebRTCDataChannelClass; struct _GstWebRTCDataChannel { GstObject parent; GstWebRTCSCTPTransport *sctp_transport; GstElement *appsrc; GstElement *appsink; gchar *label; gboolean ordered; guint max_packet_lifetime; guint max_retransmits; gchar *protocol; gboolean negotiated; gint id; GstWebRTCPriorityType priority; GstWebRTCDataChannelState ready_state; guint64 buffered_amount; guint64 buffered_amount_low_threshold; GstWebRTCBin *webrtcbin; gboolean opened; gulong src_probe; GError *stored_error; gpointer _padding[GST_PADDING]; }; struct _GstWebRTCDataChannelClass { GstObjectClass parent_class; gpointer _padding[GST_PADDING]; }; void gst_webrtc_data_channel_start_negotiation (GstWebRTCDataChannel *channel); G_GNUC_INTERNAL void gst_webrtc_data_channel_link_to_sctp (GstWebRTCDataChannel *channel, GstWebRTCSCTPTransport *sctp_transport); G_END_DECLS #endif /* __GST_WEBRTC_DATA_CHANNEL_H__ */