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729b1e9941
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`
75 lines
3.4 KiB
C++
75 lines
3.4 KiB
C++
/*************************************************************************/
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/* webrtc_peer_connection.h */
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/*************************************************************************/
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/* This file is part of: */
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/* GODOT ENGINE */
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/* https://godotengine.org */
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/*************************************************************************/
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/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */
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/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */
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/* */
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/* Permission is hereby granted, free of charge, to any person obtaining */
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/* a copy of this software and associated documentation files (the */
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/* "Software"), to deal in the Software without restriction, including */
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/* without limitation the rights to use, copy, modify, merge, publish, */
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/* distribute, sublicense, and/or sell copies of the Software, and to */
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/* permit persons to whom the Software is furnished to do so, subject to */
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/* the following conditions: */
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/* */
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/* The above copyright notice and this permission notice shall be */
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/* included in all copies or substantial portions of the Software. */
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/* */
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/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, */
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/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF */
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/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
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/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY */
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/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. */
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/*************************************************************************/
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#ifndef WEBRTC_PEER_CONNECTION_H
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#define WEBRTC_PEER_CONNECTION_H
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#include "core/io/packet_peer.h"
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#include "modules/webrtc/webrtc_data_channel.h"
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class WebRTCPeerConnection : public Reference {
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GDCLASS(WebRTCPeerConnection, Reference);
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public:
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enum ConnectionState {
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STATE_NEW,
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STATE_CONNECTING,
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STATE_CONNECTED,
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STATE_DISCONNECTED,
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STATE_FAILED,
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STATE_CLOSED
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};
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protected:
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static void _bind_methods();
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static WebRTCPeerConnection *(*_create)();
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public:
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virtual ConnectionState get_connection_state() const = 0;
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virtual Error initialize(Dictionary p_config = Dictionary()) = 0;
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virtual Ref<WebRTCDataChannel> create_data_channel(String p_label, Dictionary p_options = Dictionary()) = 0;
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virtual Error create_offer() = 0;
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virtual Error set_remote_description(String type, String sdp) = 0;
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virtual Error set_local_description(String type, String sdp) = 0;
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virtual Error add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName) = 0;
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virtual Error poll() = 0;
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virtual void close() = 0;
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static Ref<WebRTCPeerConnection> create_ref();
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static WebRTCPeerConnection *create();
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WebRTCPeerConnection();
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~WebRTCPeerConnection();
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};
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VARIANT_ENUM_CAST(WebRTCPeerConnection::ConnectionState);
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#endif // WEBRTC_PEER_CONNECTION_H
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