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/*************************************************************************/
/* 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<uint8_t> 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
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