godot/modules/webrtc/webrtc_data_channel_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

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3.7 KiB
C++

/*************************************************************************/
/* webrtc_data_channel_js.h */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
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/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */
/* */
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/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE */
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/*************************************************************************/
#ifdef JAVASCRIPT_ENABLED
#ifndef WEBRTC_DATA_CHANNEL_JS_H
#define WEBRTC_DATA_CHANNEL_JS_H
#include "webrtc_data_channel.h"
class WebRTCDataChannelJS : public WebRTCDataChannel {
GDCLASS(WebRTCDataChannelJS, WebRTCDataChannel);
private:
String _label;
String _protocol;
bool _was_string;
WriteMode _write_mode;
enum {
PACKET_BUFFER_SIZE = 65536 - 5 // 4 bytes for the size, 1 for for type
};
int _js_id;
RingBuffer<uint8_t> in_buffer;
int queue_count;
uint8_t packet_buffer[PACKET_BUFFER_SIZE];
public:
void _on_open();
void _on_close();
void _on_error();
void _on_message(uint8_t *p_data, uint32_t p_size, bool p_is_string);
virtual void set_write_mode(WriteMode mode);
virtual WriteMode get_write_mode() const;
virtual bool was_string_packet() const;
virtual ChannelState get_ready_state() const;
virtual String get_label() const;
virtual bool is_ordered() const;
virtual int get_id() const;
virtual int get_max_packet_life_time() const;
virtual int get_max_retransmits() const;
virtual String get_protocol() const;
virtual bool is_negotiated() const;
virtual Error poll();
virtual void close();
/** 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;
WebRTCDataChannelJS();
WebRTCDataChannelJS(int js_id);
~WebRTCDataChannelJS();
};
#endif // WEBRTC_DATA_CHANNEL_JS_H
#endif // JAVASCRIPT_ENABLED