godot/modules/webrtc/webrtc_peer_connection.h
Fabio Alessandrelli dbff58ebdd [WebRTC] Expose more of the WebRTC API.
Add get_gathering_state() returning the iceGatheringState of the
connection.

Add get_signaling_state() returning the signalingState of the
connection.

Improve JS library.
2022-09-13 14:46:58 +02:00

98 lines
4.0 KiB
C++

/*************************************************************************/
/* webrtc_peer_connection.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2022 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2022 Godot Engine contributors (cf. AUTHORS.md). */
/* */
/* Permission is hereby granted, free of charge, to any person obtaining */
/* a copy of this software and associated documentation files (the */
/* "Software"), to deal in the Software without restriction, including */
/* without limitation the rights to use, copy, modify, merge, publish, */
/* distribute, sublicense, and/or sell copies of the Software, and to */
/* permit persons to whom the Software is furnished to do so, subject to */
/* the following conditions: */
/* */
/* The above copyright notice and this permission notice shall be */
/* included in all copies or substantial portions of the Software. */
/* */
/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/*************************************************************************/
#ifndef WEBRTC_PEER_CONNECTION_H
#define WEBRTC_PEER_CONNECTION_H
#include "core/io/packet_peer.h"
#include "modules/webrtc/webrtc_data_channel.h"
class WebRTCPeerConnection : public RefCounted {
GDCLASS(WebRTCPeerConnection, RefCounted);
public:
enum ConnectionState {
STATE_NEW,
STATE_CONNECTING,
STATE_CONNECTED,
STATE_DISCONNECTED,
STATE_FAILED,
STATE_CLOSED
};
enum GatheringState {
GATHERING_STATE_NEW,
GATHERING_STATE_GATHERING,
GATHERING_STATE_COMPLETE,
};
enum SignalingState {
SIGNALING_STATE_STABLE,
SIGNALING_STATE_HAVE_LOCAL_OFFER,
SIGNALING_STATE_HAVE_REMOTE_OFFER,
SIGNALING_STATE_HAVE_LOCAL_PRANSWER,
SIGNALING_STATE_HAVE_REMOTE_PRANSWER,
SIGNALING_STATE_CLOSED,
};
private:
static StringName default_extension;
protected:
static void _bind_methods();
public:
static void set_default_extension(const StringName &p_name);
virtual ConnectionState get_connection_state() const = 0;
virtual GatheringState get_gathering_state() const = 0;
virtual SignalingState get_signaling_state() const = 0;
virtual Error initialize(Dictionary p_config = Dictionary()) = 0;
virtual Ref<WebRTCDataChannel> create_data_channel(String p_label, Dictionary p_options = Dictionary()) = 0;
virtual Error create_offer() = 0;
virtual Error set_remote_description(String type, String sdp) = 0;
virtual Error set_local_description(String type, String sdp) = 0;
virtual Error add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName) = 0;
virtual Error poll() = 0;
virtual void close() = 0;
static WebRTCPeerConnection *create();
WebRTCPeerConnection();
~WebRTCPeerConnection();
};
VARIANT_ENUM_CAST(WebRTCPeerConnection::ConnectionState);
VARIANT_ENUM_CAST(WebRTCPeerConnection::GatheringState);
VARIANT_ENUM_CAST(WebRTCPeerConnection::SignalingState);
#endif // WEBRTC_PEER_CONNECTION_H