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godot/modules/webrtc/webrtc_data_channel_gdnative.h

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One Copyright Update to rule them all As many open source projects have started doing it, we're removing the current year from the copyright notice, so that we don't need to bump it every year. It seems like only the first year of publication is technically relevant for copyright notices, and even that seems to be something that many companies stopped listing altogether (in a version controlled codebase, the commits are a much better source of date of publication than a hardcoded copyright statement). We also now list Godot Engine contributors first as we're collectively the current maintainers of the project, and we clarify that the "exclusive" copyright of the co-founders covers the timespan before opensourcing (their further contributions are included as part of Godot Engine contributors). Also fixed "cf." Frenchism - it's meant as "refer to / see". Backported from #70885.
2023-01-10 14:26:54 +00:00
/**************************************************************************/
/* webrtc_data_channel_gdnative.h */
/**************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/**************************************************************************/
/* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */
/* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */
/* */
/* 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. */
/**************************************************************************/
Add WebRTC GDNative interface
2018-11-20 09:58:56 +00:00
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`
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#ifndef WEBRTC_DATA_CHANNEL_GDNATIVE_H
#define WEBRTC_DATA_CHANNEL_GDNATIVE_H
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Ensure header guards enclose entire header. (cherry picked from commit f6ad1954f76165aefb05e12e5f82cbf5d763c85e)
2020-09-03 10:28:55 +00:00
#ifdef WEBRTC_GDNATIVE_ENABLED
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#include "modules/gdnative/include/net/godot_net.h"
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`
2019-05-10 23:46:27 +00:00
#include "webrtc_data_channel.h"
Add WebRTC GDNative interface
2018-11-20 09:58:56 +00:00
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`
2019-05-10 23:46:27 +00:00
class WebRTCDataChannelGDNative : public WebRTCDataChannel {
GDCLASS(WebRTCDataChannelGDNative, WebRTCDataChannel);
Add WebRTC GDNative interface
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protected:
static void _bind_methods();
private:
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`
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const godot_net_webrtc_data_channel *interface;
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public:
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`
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void set_native_webrtc_data_channel(const godot_net_webrtc_data_channel *p_impl);
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virtual void set_write_mode(WriteMode mode);
virtual WriteMode get_write_mode() const;
virtual bool was_string_packet() const;
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`
2019-05-10 23:46:27 +00:00
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;
Add get_buffered_amount() to WebRTCDataChannel
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virtual int get_buffered_amount() const;
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`
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Add WebRTC GDNative interface
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virtual Error poll();
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`
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virtual void close();
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/** 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;
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`
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WebRTCDataChannelGDNative();
~WebRTCDataChannelGDNative();
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};
#endif // WEBRTC_GDNATIVE_ENABLED
Ensure header guards enclose entire header. (cherry picked from commit f6ad1954f76165aefb05e12e5f82cbf5d763c85e)
2020-09-03 10:28:55 +00:00
#endif // WEBRTC_DATA_CHANNEL_GDNATIVE_H