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729b1e9941
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`
85 lines
3.6 KiB
C++
85 lines
3.6 KiB
C++
/*************************************************************************/
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/* webrtc_data_channel.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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#ifndef WEBRTC_DATA_CHANNEL_H
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#define WEBRTC_DATA_CHANNEL_H
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#include "core/io/packet_peer.h"
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class WebRTCDataChannel : public PacketPeer {
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GDCLASS(WebRTCDataChannel, PacketPeer);
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public:
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enum WriteMode {
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WRITE_MODE_TEXT,
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WRITE_MODE_BINARY,
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};
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enum ChannelState {
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STATE_CONNECTING,
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STATE_OPEN,
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STATE_CLOSING,
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STATE_CLOSED
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};
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protected:
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static void _bind_methods();
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public:
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virtual void set_write_mode(WriteMode mode) = 0;
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virtual WriteMode get_write_mode() const = 0;
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virtual bool was_string_packet() const = 0;
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virtual ChannelState get_ready_state() const = 0;
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virtual String get_label() const = 0;
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virtual bool is_ordered() const = 0;
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virtual int get_id() const = 0;
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virtual int get_max_packet_life_time() const = 0;
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virtual int get_max_retransmits() const = 0;
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virtual String get_protocol() const = 0;
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virtual bool is_negotiated() const = 0;
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virtual Error poll() = 0;
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virtual void close() = 0;
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/** Inherited from PacketPeer: **/
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virtual int get_available_packet_count() const = 0;
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virtual Error get_packet(const uint8_t **r_buffer, int &r_buffer_size) = 0; ///< buffer is GONE after next get_packet
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virtual Error put_packet(const uint8_t *p_buffer, int p_buffer_size) = 0;
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virtual int get_max_packet_size() const = 0;
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WebRTCDataChannel();
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~WebRTCDataChannel();
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};
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VARIANT_ENUM_CAST(WebRTCDataChannel::WriteMode);
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VARIANT_ENUM_CAST(WebRTCDataChannel::ChannelState);
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#endif // WEBRTC_DATA_CHANNEL_H
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