godot/modules/webrtc/webrtc_peer_connection_js.h
Fabio Alessandrelli 729b1e9941 WebRTC refactor. Data channels, STUN/TURN support.
A big refactor to the WebRTC module. API is now considered quite stable.

Highlights:

- Renamed `WebRTCPeer` to `WebRTCPeerConnection`.
- `WebRTCPeerConnection` no longer act as `PacketPeer`, it only handle the connection itself (a bit like `TCP_Server`)
- Added new `WebRTCDataChannel` class which inherits from `PacketPeer` to handle data transfer.
- Add `WebRTCPeerConnection.initialize` method to create a new connection with the desired configuration provided as dictionary ([see MDN docs](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#RTCConfiguration_dictionary)).
- Add `WebRTCPeerConnection.create_data_channel` method to create a data channel for the given connection. The connection must be in `STATE_NEW` as specified by the standard ([see MDN docs for options](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createDataChannel#RTCDataChannelInit_dictionary)).
- Add a `data_channel_received` signal to `WebRTCPeerConnection` for in-band (not negotiated) channels.
- Renamed `WebRTCPeerConnection` `offer_created` signal to `session_description_created`.
- Renamed `WebRTCPeerConnection` `new_ice_candidate` signal to `ice_candidate_created`
2019-05-16 11:21:20 +02:00

67 lines
3.3 KiB
C++

/*************************************************************************/
/* webrtc_peer_connection_js.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2019 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 */
/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
/*************************************************************************/
#ifndef WEBRTC_PEER_CONNECTION_JS_H
#define WEBRTC_PEER_CONNECTION_JS_H
#ifdef JAVASCRIPT_ENABLED
#include "webrtc_peer_connection.h"
class WebRTCPeerConnectionJS : public WebRTCPeerConnection {
private:
int _js_id;
ConnectionState _conn_state;
public:
static WebRTCPeerConnection *_create() { return memnew(WebRTCPeerConnectionJS); }
static void make_default() { WebRTCPeerConnection::_create = WebRTCPeerConnectionJS::_create; }
void _on_connection_state_changed();
virtual ConnectionState get_connection_state() const;
virtual Error initialize(Dictionary configuration = Dictionary());
virtual Ref<WebRTCDataChannel> create_data_channel(String p_channel_name, Dictionary p_channel_config = Dictionary());
virtual Error create_offer();
virtual Error set_remote_description(String type, String sdp);
virtual Error set_local_description(String type, String sdp);
virtual Error add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName);
virtual Error poll();
virtual void close();
WebRTCPeerConnectionJS();
~WebRTCPeerConnectionJS();
};
#endif
#endif // WEBRTC_PEER_CONNECTION_JS_H