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95 lines
3.8 KiB
C
95 lines
3.8 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_WEBRTC_RTP_TRANSCEIVER_H__
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#define __GST_WEBRTC_RTP_TRANSCEIVER_H__
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#include <gst/gst.h>
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#include <gst/webrtc/webrtc_fwd.h>
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#include <gst/webrtc/rtpsender.h>
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#include <gst/webrtc/rtpreceiver.h>
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G_BEGIN_DECLS
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GST_WEBRTC_API
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GType gst_webrtc_rtp_transceiver_get_type(void);
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#define GST_TYPE_WEBRTC_RTP_TRANSCEIVER (gst_webrtc_rtp_transceiver_get_type())
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#define GST_WEBRTC_RTP_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiver))
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#define GST_IS_WEBRTC_RTP_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_RTP_TRANSCEIVER))
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#define GST_WEBRTC_RTP_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
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#define GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER))
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#define GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
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/**
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* GstWebRTCRTPTransceiver:
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* @mline: the mline number this transceiver corresponds to
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* @mid: The media ID of the m-line associated with this
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* transceiver. This association is established, when possible,
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* whenever either a local or remote description is applied. This
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* field is NULL if neither a local or remote description has been
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* applied, or if its associated m-line is rejected by either a remote
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* offer or any answer.
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* @stopped: Indicates whether or not sending and receiving using the paired
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* #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
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* either due to SDP offer/answer
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* @sender: The #GstWebRTCRTPSender object responsible sending data to the
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* remote peer
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* @receiver: The #GstWebRTCRTPReceiver object responsible for receiver data from
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* the remote peer.
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* @direction: The transceiver's desired direction.
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* @current_direction: The transceiver's current direction (read-only)
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* @codec_preferences: A caps representing the codec preferences (read-only)
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* @kind: Type of media (Since: 1.20)
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*
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* Mostly matches the WebRTC RTCRtpTransceiver interface.
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*
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* Since: 1.16
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*/
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struct _GstWebRTCRTPTransceiver
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{
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GstObject parent;
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guint mline;
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gchar *mid;
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gboolean stopped;
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GstWebRTCRTPSender *sender;
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GstWebRTCRTPReceiver *receiver;
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GstWebRTCRTPTransceiverDirection direction;
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GstWebRTCRTPTransceiverDirection current_direction;
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GstCaps *codec_preferences;
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GstWebRTCKind kind;
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gpointer _padding[GST_PADDING];
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};
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struct _GstWebRTCRTPTransceiverClass
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{
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GstObjectClass parent_class;
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/* FIXME; reset */
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gpointer _padding[GST_PADDING];
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};
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G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPTransceiver, gst_object_unref)
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G_END_DECLS
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#endif /* __GST_WEBRTC_RTP_TRANSCEIVER_H__ */
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