gstreamer/subprojects/gst-plugins-bad/gst-libs/gst/webrtc/webrtc-priv.h

274 lines
7.7 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_PRIV_H__
#define __GST_WEBRTC_PRIV_H__
#include <gst/gst.h>
#include <gst/webrtc/webrtc_fwd.h>
#include <gst/webrtc/rtpsender.h>
#include <gst/webrtc/rtpreceiver.h>
G_BEGIN_DECLS
/**
* 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.
*/
/**
* 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];
};
/**
* GstWebRTCRTPSender:
* @transport: The transport for RTP packets
* @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.
*/
/**
* 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 as necessary */
GstWebRTCDTLSTransport *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);
/**
* GstWebRTCRTPReceiver:
* @transport: The transport for RTP packets
*
* An object to track the receiving aspect of the stream
*
* Mostly matches the WebRTC RTCRtpReceiver interface.
*/
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);
/**
* GstWebRTCDTLSTransport:
*/
struct _GstWebRTCDTLSTransport
{
GstObject parent;
GstWebRTCICETransport *transport;
GstWebRTCDTLSTransportState state;
gboolean client;
guint session_id;
GstElement *dtlssrtpenc;
GstElement *dtlssrtpdec;
gpointer _padding[GST_PADDING];
};
struct _GstWebRTCDTLSTransportClass
{
GstObjectClass parent_class;
gpointer _padding[GST_PADDING];
};
GST_WEBRTC_API
GstWebRTCDTLSTransport * gst_webrtc_dtls_transport_new (guint session_id);
GST_WEBRTC_API
void gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
GstWebRTCICETransport * ice);
#define GST_WEBRTC_DATA_CHANNEL_LOCK(channel) g_mutex_lock(&((GstWebRTCDataChannel *)(channel))->lock)
#define GST_WEBRTC_DATA_CHANNEL_UNLOCK(channel) g_mutex_unlock(&((GstWebRTCDataChannel *)(channel))->lock)
/**
* GstWebRTCDataChannel:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannel
{
GObject parent;
GMutex lock;
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;
gpointer _padding[GST_PADDING];
};
/**
* GstWebRTCDataChannelClass:
*
* Since: 1.18
*/
struct _GstWebRTCDataChannelClass
{
GObjectClass parent_class;
void (*send_data) (GstWebRTCDataChannel * channel, GBytes *data);
void (*send_string) (GstWebRTCDataChannel * channel, const gchar *str);
void (*close) (GstWebRTCDataChannel * channel);
gpointer _padding[GST_PADDING];
};
GST_WEBRTC_API
void gst_webrtc_data_channel_on_open (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_close (GstWebRTCDataChannel * channel);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_error (GstWebRTCDataChannel * channel, GError * error);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_data (GstWebRTCDataChannel * channel, GBytes * data);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_message_string (GstWebRTCDataChannel * channel, const gchar * str);
GST_WEBRTC_API
void gst_webrtc_data_channel_on_buffered_amount_low (GstWebRTCDataChannel * channel);
/**
* GstWebRTCSCTPTransport:
*
* Since: 1.20
*/
struct _GstWebRTCSCTPTransport
{
GstObject parent;
};
/**
* GstWebRTCSCTPTransportClass:
*
* Since: 1.20
*/
struct _GstWebRTCSCTPTransportClass
{
GstObjectClass parent_class;
};
G_END_DECLS
#endif /* __GST_WEBRTC_PRIV_H__ */