mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-22 16:26:39 +00:00
7be09a5f22
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
102 lines
3.9 KiB
C
102 lines
3.9 KiB
C
/* 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_TRANSCEIVER_H__
|
|
#define __GST_WEBRTC_RTP_TRANSCEIVER_H__
|
|
|
|
#include <gst/gst.h>
|
|
#include <gst/webrtc/webrtc_fwd.h>
|
|
#include <gst/webrtc/rtpsender.h>
|
|
#include <gst/webrtc/rtpreceiver.h>
|
|
|
|
G_BEGIN_DECLS
|
|
|
|
GST_WEBRTC_API
|
|
GType gst_webrtc_rtp_transceiver_get_type(void);
|
|
#define GST_TYPE_WEBRTC_RTP_TRANSCEIVER (gst_webrtc_rtp_transceiver_get_type())
|
|
#define GST_WEBRTC_RTP_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiver))
|
|
#define GST_IS_WEBRTC_RTP_TRANSCEIVER(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_WEBRTC_RTP_TRANSCEIVER))
|
|
#define GST_WEBRTC_RTP_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
|
|
#define GST_IS_WEBRTC_RTP_TRANSCEIVER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER))
|
|
#define GST_WEBRTC_RTP_TRANSCEIVER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_TRANSCEIVER,GstWebRTCRTPTransceiverClass))
|
|
|
|
/**
|
|
* GstWebRTCRTPTransceiver:
|
|
* @mline: the mline number this transceiver corresponds to
|
|
* @mid: The media ID of the m-line associated with this
|
|
* transceiver. This association is established, when possible,
|
|
* whenever either a local or remote description is applied. This
|
|
* field is NULL if neither a local or remote description has been
|
|
* applied, or if its associated m-line is rejected by either a remote
|
|
* offer or any answer.
|
|
* @stopped: Indicates whether or not sending and receiving using the paired
|
|
* #GstWebRTCRTPSender and #GstWebRTCRTPReceiver has been permanently disabled,
|
|
* either due to SDP offer/answer
|
|
* @sender: The #GstWebRTCRTPSender object responsible sending data to the
|
|
* remote peer
|
|
* @receiver: The #GstWebRTCRTPReceiver object responsible for receiver data from
|
|
* the remote peer.
|
|
* @direction: The transceiver's desired direction.
|
|
* @current_direction: The transceiver's current direction (read-only)
|
|
* @codec_preferences: A caps representing the codec preferences (read-only)
|
|
* @kind: Type of media (Since: 1.20)
|
|
*
|
|
* Mostly matches the WebRTC RTCRtpTransceiver interface.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
/**
|
|
* GstWebRTCRTPTransceiver.kind:
|
|
*
|
|
* Type of media
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
struct _GstWebRTCRTPTransceiver
|
|
{
|
|
GstObject parent;
|
|
guint mline;
|
|
gchar *mid;
|
|
gboolean stopped;
|
|
|
|
GstWebRTCRTPSender *sender;
|
|
GstWebRTCRTPReceiver *receiver;
|
|
|
|
GstWebRTCRTPTransceiverDirection direction;
|
|
GstWebRTCRTPTransceiverDirection current_direction;
|
|
|
|
GstCaps *codec_preferences;
|
|
GstWebRTCKind kind;
|
|
|
|
gpointer _padding[GST_PADDING];
|
|
};
|
|
|
|
struct _GstWebRTCRTPTransceiverClass
|
|
{
|
|
GstObjectClass parent_class;
|
|
|
|
/* FIXME; reset */
|
|
gpointer _padding[GST_PADDING];
|
|
};
|
|
|
|
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPTransceiver, gst_object_unref)
|
|
|
|
G_END_DECLS
|
|
|
|
#endif /* __GST_WEBRTC_RTP_TRANSCEIVER_H__ */
|