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	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`
		
			
				
	
	
		
			93 lines
		
	
	
	
		
			3.7 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
			
		
		
	
	
			93 lines
		
	
	
	
		
			3.7 KiB
		
	
	
	
		
			C++
		
	
	
	
	
	
/*************************************************************************/
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/*  webrtc_data_channel_js.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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#ifdef JAVASCRIPT_ENABLED
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#ifndef WEBRTC_DATA_CHANNEL_JS_H
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#define WEBRTC_DATA_CHANNEL_JS_H
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#include "webrtc_data_channel.h"
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class WebRTCDataChannelJS : public WebRTCDataChannel {
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	GDCLASS(WebRTCDataChannelJS, WebRTCDataChannel);
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private:
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	String _label;
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	String _protocol;
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	bool _was_string;
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	WriteMode _write_mode;
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	enum {
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		PACKET_BUFFER_SIZE = 65536 - 5 // 4 bytes for the size, 1 for for type
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	};
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	int _js_id;
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	RingBuffer<uint8_t> in_buffer;
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	int queue_count;
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	uint8_t packet_buffer[PACKET_BUFFER_SIZE];
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public:
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	void _on_open();
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	void _on_close();
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	void _on_error();
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	void _on_message(uint8_t *p_data, uint32_t p_size, bool p_is_string);
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	virtual void set_write_mode(WriteMode mode);
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	virtual WriteMode get_write_mode() const;
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	virtual bool was_string_packet() const;
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	virtual ChannelState get_ready_state() const;
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	virtual String get_label() const;
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	virtual bool is_ordered() const;
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	virtual int get_id() const;
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	virtual int get_max_packet_life_time() const;
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	virtual int get_max_retransmits() const;
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	virtual String get_protocol() const;
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	virtual bool is_negotiated() const;
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	virtual Error poll();
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	virtual void close();
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	/** Inherited from PacketPeer: **/
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	virtual int get_available_packet_count() const;
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	virtual Error get_packet(const uint8_t **r_buffer, int &r_buffer_size); ///< buffer is GONE after next get_packet
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	virtual Error put_packet(const uint8_t *p_buffer, int p_buffer_size);
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	virtual int get_max_packet_size() const;
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	WebRTCDataChannelJS();
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	WebRTCDataChannelJS(int js_id);
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	~WebRTCDataChannelJS();
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};
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#endif // WEBRTC_DATA_CHANNEL_JS_H
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#endif // JAVASCRIPT_ENABLED
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