godot/modules/webrtc/webrtc_data_channel.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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3.6 KiB
C++

/*************************************************************************/
/* webrtc_data_channel.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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/*************************************************************************/
#ifndef WEBRTC_DATA_CHANNEL_H
#define WEBRTC_DATA_CHANNEL_H
#include "core/io/packet_peer.h"
class WebRTCDataChannel : public PacketPeer {
GDCLASS(WebRTCDataChannel, PacketPeer);
public:
enum WriteMode {
WRITE_MODE_TEXT,
WRITE_MODE_BINARY,
};
enum ChannelState {
STATE_CONNECTING,
STATE_OPEN,
STATE_CLOSING,
STATE_CLOSED
};
protected:
static void _bind_methods();
public:
virtual void set_write_mode(WriteMode mode) = 0;
virtual WriteMode get_write_mode() const = 0;
virtual bool was_string_packet() const = 0;
virtual ChannelState get_ready_state() const = 0;
virtual String get_label() const = 0;
virtual bool is_ordered() const = 0;
virtual int get_id() const = 0;
virtual int get_max_packet_life_time() const = 0;
virtual int get_max_retransmits() const = 0;
virtual String get_protocol() const = 0;
virtual bool is_negotiated() const = 0;
virtual Error poll() = 0;
virtual void close() = 0;
/** Inherited from PacketPeer: **/
virtual int get_available_packet_count() const = 0;
virtual Error get_packet(const uint8_t **r_buffer, int &r_buffer_size) = 0; ///< buffer is GONE after next get_packet
virtual Error put_packet(const uint8_t *p_buffer, int p_buffer_size) = 0;
virtual int get_max_packet_size() const = 0;
WebRTCDataChannel();
~WebRTCDataChannel();
};
VARIANT_ENUM_CAST(WebRTCDataChannel::WriteMode);
VARIANT_ENUM_CAST(WebRTCDataChannel::ChannelState);
#endif // WEBRTC_DATA_CHANNEL_H