/*************************************************************************/ /* webrtc_peer_js.h */ /*************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* https://godotengine.org */ /*************************************************************************/ /* Copyright (c) 2007-2018 Juan Linietsky, Ariel Manzur. */ /* Copyright (c) 2014-2018 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_JS_H #define WEBRTC_PEER_JS_H #ifdef JAVASCRIPT_ENABLED #include "webrtc_peer.h" class WebRTCPeerJS : public WebRTCPeer { private: enum { PACKET_BUFFER_SIZE = 65536 - 5 // 4 bytes for the size, 1 for for type }; bool _was_string; WriteMode _write_mode; int _js_id; RingBuffer in_buffer; int queue_count; uint8_t packet_buffer[PACKET_BUFFER_SIZE]; ConnectionState _conn_state; public: static WebRTCPeer *_create() { return memnew(WebRTCPeerJS); } static void make_default() { WebRTCPeer::_create = WebRTCPeerJS::_create; } virtual void set_write_mode(WriteMode mode); virtual WriteMode get_write_mode() const; virtual bool was_string_packet() const; virtual ConnectionState get_connection_state() const; 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(); /** Inherited from PacketPeer: **/ virtual int get_available_packet_count() const; virtual Error get_packet(const uint8_t **r_buffer, int &r_buffer_size); ///< buffer is GONE after next get_packet virtual Error put_packet(const uint8_t *p_buffer, int p_buffer_size); virtual int get_max_packet_size() const; void close(); void _on_open(); void _on_close(); void _on_error(); void _on_message(uint8_t *p_data, uint32_t p_size, bool p_is_string); WebRTCPeerJS(); ~WebRTCPeerJS(); }; #endif #endif // WEBRTC_PEER_JS_H