diff options
Diffstat (limited to 'modules/webrtc')
21 files changed, 2577 insertions, 0 deletions
diff --git a/modules/webrtc/SCsub b/modules/webrtc/SCsub new file mode 100644 index 0000000000..868553b879 --- /dev/null +++ b/modules/webrtc/SCsub @@ -0,0 +1,15 @@ +#!/usr/bin/env python + +Import('env') +Import('env_modules') + +# Thirdparty source files + +env_webrtc = env_modules.Clone() +use_gdnative = env_webrtc["module_gdnative_enabled"] + +if use_gdnative: # GDNative is retained in Javascript for export compatibility + env_webrtc.Append(CPPDEFINES=['WEBRTC_GDNATIVE_ENABLED']) + env_webrtc.Prepend(CPPPATH=["#modules/gdnative/include/"]) + +env_webrtc.add_source_files(env.modules_sources, "*.cpp") diff --git a/modules/webrtc/config.py b/modules/webrtc/config.py new file mode 100644 index 0000000000..48b4c33c5d --- /dev/null +++ b/modules/webrtc/config.py @@ -0,0 +1,15 @@ +def can_build(env, platform): + return True + +def configure(env): + pass + +def get_doc_classes(): + return [ + "WebRTCPeerConnection", + "WebRTCDataChannel", + "WebRTCMultiplayer" + ] + +def get_doc_path(): + return "doc_classes" diff --git a/modules/webrtc/doc_classes/WebRTCDataChannel.xml b/modules/webrtc/doc_classes/WebRTCDataChannel.xml new file mode 100644 index 0000000000..98715ee99b --- /dev/null +++ b/modules/webrtc/doc_classes/WebRTCDataChannel.xml @@ -0,0 +1,116 @@ +<?xml version="1.0" encoding="UTF-8" ?> +<class name="WebRTCDataChannel" inherits="PacketPeer" category="Core" version="3.2"> + <brief_description> + </brief_description> + <description> + </description> + <tutorials> + </tutorials> + <methods> + <method name="close"> + <return type="void"> + </return> + <description> + Closes this data channel, notifying the other peer. + </description> + </method> + <method name="get_id" qualifiers="const"> + <return type="int"> + </return> + <description> + Returns the id assigned to this channel during creation (or auto-assigned during negotiation). + If the channel is not negotiated out-of-band the id will only be available after the connection is established (will return [code]65535[/code] until then). + </description> + </method> + <method name="get_label" qualifiers="const"> + <return type="String"> + </return> + <description> + Returns the label assigned to this channel during creation. + </description> + </method> + <method name="get_max_packet_life_time" qualifiers="const"> + <return type="int"> + </return> + <description> + Returns the [code]maxPacketLifeTime[/code] value assigned to this channel during creation. + Will be [code]65535[/code] if not specified. + </description> + </method> + <method name="get_max_retransmits" qualifiers="const"> + <return type="int"> + </return> + <description> + Returns the [code]maxRetransmits[/code] value assigned to this channel during creation. + Will be [code]65535[/code] if not specified. + </description> + </method> + <method name="get_protocol" qualifiers="const"> + <return type="String"> + </return> + <description> + Returns the sub-protocol assigned to this channel during creation. An empty string if not specified. + </description> + </method> + <method name="get_ready_state" qualifiers="const"> + <return type="int" enum="WebRTCDataChannel.ChannelState"> + </return> + <description> + Returns the current state of this channel, see [enum ChannelState]. + </description> + </method> + <method name="is_negotiated" qualifiers="const"> + <return type="bool"> + </return> + <description> + Returns [code]true[/code] if this channel was created with out-of-band configuration. + </description> + </method> + <method name="is_ordered" qualifiers="const"> + <return type="bool"> + </return> + <description> + Returns [code]true[/code] if this channel was created with ordering enabled (default). + </description> + </method> + <method name="poll"> + <return type="int" enum="Error"> + </return> + <description> + Reserved, but not used for now. + </description> + </method> + <method name="was_string_packet" qualifiers="const"> + <return type="bool"> + </return> + <description> + Returns [code]true[/code] if the last received packet was transferred as text. See [member write_mode]. + </description> + </method> + </methods> + <members> + <member name="write_mode" type="int" setter="set_write_mode" getter="get_write_mode" enum="WebRTCDataChannel.WriteMode" default="1"> + The transfer mode to use when sending outgoing packet. Either text or binary. + </member> + </members> + <constants> + <constant name="WRITE_MODE_TEXT" value="0" enum="WriteMode"> + Tells the channel to send data over this channel as text. An external peer (non-Godot) would receive this as a string. + </constant> + <constant name="WRITE_MODE_BINARY" value="1" enum="WriteMode"> + Tells the channel to send data over this channel as binary. An external peer (non-Godot) would receive this as array buffer or blob. + </constant> + <constant name="STATE_CONNECTING" value="0" enum="ChannelState"> + The channel was created, but it's still trying to connect. + </constant> + <constant name="STATE_OPEN" value="1" enum="ChannelState"> + The channel is currently open, and data can flow over it. + </constant> + <constant name="STATE_CLOSING" value="2" enum="ChannelState"> + The channel is being closed, no new messages will be accepted, but those already in queue will be flushed. + </constant> + <constant name="STATE_CLOSED" value="3" enum="ChannelState"> + The channel was closed, or connection failed. + </constant> + </constants> +</class> diff --git a/modules/webrtc/doc_classes/WebRTCMultiplayer.xml b/modules/webrtc/doc_classes/WebRTCMultiplayer.xml new file mode 100644 index 0000000000..605b1ef082 --- /dev/null +++ b/modules/webrtc/doc_classes/WebRTCMultiplayer.xml @@ -0,0 +1,89 @@ +<?xml version="1.0" encoding="UTF-8" ?> +<class name="WebRTCMultiplayer" inherits="NetworkedMultiplayerPeer" category="Core" version="3.2"> + <brief_description> + A simple interface to create a peer-to-peer mesh network composed of [WebRTCPeerConnection] that is compatible with the [MultiplayerAPI]. + </brief_description> + <description> + This class constructs a full mesh of [WebRTCPeerConnection] (one connection for each peer) that can be used as a [member MultiplayerAPI.network_peer]. + You can add each [WebRTCPeerConnection] via [method add_peer] or remove them via [method remove_peer]. Peers must be added in [constant WebRTCPeerConnection.STATE_NEW] state to allow it to create the appropriate channels. This class will not create offers nor set descriptions, it will only poll them, and notify connections and disconnections. + [signal NetworkedMultiplayerPeer.connection_succeeded] and [signal NetworkedMultiplayerPeer.server_disconnected] will not be emitted unless [code]server_compatibility[/code] is [code]true[/code] in [method initialize]. Beside that data transfer works like in a [NetworkedMultiplayerPeer]. + </description> + <tutorials> + </tutorials> + <methods> + <method name="add_peer"> + <return type="int" enum="Error"> + </return> + <argument index="0" name="peer" type="WebRTCPeerConnection"> + </argument> + <argument index="1" name="peer_id" type="int"> + </argument> + <argument index="2" name="unreliable_lifetime" type="int" default="1"> + </argument> + <description> + Add a new peer to the mesh with the given [code]peer_id[/code]. The [WebRTCPeerConnection] must be in state [constant WebRTCPeerConnection.STATE_NEW]. + Three channels will be created for reliable, unreliable, and ordered transport. The value of [code]unreliable_lifetime[/code] will be passed to the [code]maxPacketLifetime[/code] option when creating unreliable and ordered channels (see [method WebRTCPeerConnection.create_data_channel]). + </description> + </method> + <method name="close"> + <return type="void"> + </return> + <description> + Close all the add peer connections and channels, freeing all resources. + </description> + </method> + <method name="get_peer"> + <return type="Dictionary"> + </return> + <argument index="0" name="peer_id" type="int"> + </argument> + <description> + Return a dictionary representation of the peer with given [code]peer_id[/code] with three keys. [code]connection[/code] containing the [WebRTCPeerConnection] to this peer, [code]channels[/code] an array of three [WebRTCDataChannel], and [code]connected[/code] a boolean representing if the peer connection is currently connected (all three channels are open). + </description> + </method> + <method name="get_peers"> + <return type="Dictionary"> + </return> + <description> + Returns a dictionary which keys are the peer ids and values the peer representation as in [method get_peer] + </description> + </method> + <method name="has_peer"> + <return type="bool"> + </return> + <argument index="0" name="peer_id" type="int"> + </argument> + <description> + Returns [code]true[/code] if the given [code]peer_id[/code] is in the peers map (it might not be connected though). + </description> + </method> + <method name="initialize"> + <return type="int" enum="Error"> + </return> + <argument index="0" name="peer_id" type="int"> + </argument> + <argument index="1" name="server_compatibility" type="bool" default="false"> + </argument> + <description> + Initialize the multiplayer peer with the given [code]peer_id[/code] (must be between 1 and 2147483647). + If [code]server_compatibilty[/code] is [code]false[/code] (default), the multiplayer peer will be immediately in state [constant NetworkedMultiplayerPeer.CONNECTION_CONNECTED] and [signal NetworkedMultiplayerPeer.connection_succeeded] will not be emitted. + If [code]server_compatibilty[/code] is [code]true[/code] the peer will suppress all [signal NetworkedMultiplayerPeer.peer_connected] signals until a peer with id [constant NetworkedMultiplayerPeer.TARGET_PEER_SERVER] connects and then emit [signal NetworkedMultiplayerPeer.connection_succeeded]. After that the signal [signal NetworkedMultiplayerPeer.peer_connected] will be emitted for every already connected peer, and any new peer that might connect. If the server peer disconnects after that, signal [signal NetworkedMultiplayerPeer.server_disconnected] will be emitted and state will become [constant NetworkedMultiplayerPeer.CONNECTION_CONNECTED]. + </description> + </method> + <method name="remove_peer"> + <return type="void"> + </return> + <argument index="0" name="peer_id" type="int"> + </argument> + <description> + Remove the peer with given [code]peer_id[/code] from the mesh. If the peer was connected, and [signal NetworkedMultiplayerPeer.peer_connected] was emitted for it, then [signal NetworkedMultiplayerPeer.peer_disconnected] will be emitted. + </description> + </method> + </methods> + <members> + <member name="refuse_new_connections" type="bool" setter="set_refuse_new_connections" getter="is_refusing_new_connections" override="true" default="false" /> + <member name="transfer_mode" type="int" setter="set_transfer_mode" getter="get_transfer_mode" override="true" enum="NetworkedMultiplayerPeer.TransferMode" default="2" /> + </members> + <constants> + </constants> +</class> diff --git a/modules/webrtc/doc_classes/WebRTCPeerConnection.xml b/modules/webrtc/doc_classes/WebRTCPeerConnection.xml new file mode 100644 index 0000000000..26082d73a8 --- /dev/null +++ b/modules/webrtc/doc_classes/WebRTCPeerConnection.xml @@ -0,0 +1,190 @@ +<?xml version="1.0" encoding="UTF-8" ?> +<class name="WebRTCPeerConnection" inherits="Reference" category="Core" version="3.2"> + <brief_description> + Interface to a WebRTC peer connection. + </brief_description> + <description> + A WebRTC connection between the local computer and a remote peer. Provides an interface to connect, maintain and monitor the connection. + Setting up a WebRTC connection between two peers from now on) may not seem a trivial task, but it can be broken down into 3 main steps: + - The peer that wants to initiate the connection ([code]A[/code] from now on) creates an offer and send it to the other peer ([code]B[/code] from now on). + - [code]B[/code] receives the offer, generate and answer, and sends it to [code]A[/code]). + - [code]A[/code] and [code]B[/code] then generates and exchange ICE candidates with each other. + After these steps, the connection should become connected. Keep on reading or look into the tutorial for more information. + </description> + <tutorials> + </tutorials> + <methods> + <method name="add_ice_candidate"> + <return type="int" enum="Error"> + </return> + <argument index="0" name="media" type="String"> + </argument> + <argument index="1" name="index" type="int"> + </argument> + <argument index="2" name="name" type="String"> + </argument> + <description> + Add an ice candidate generated by a remote peer (and received over the signaling server). See [signal ice_candidate_created]. + </description> + </method> + <method name="close"> + <return type="void"> + </return> + <description> + Close the peer connection and all data channels associated with it. Note, you cannot reuse this object for a new connection unless you call [method initialize]. + </description> + </method> + <method name="create_data_channel"> + <return type="WebRTCDataChannel"> + </return> + <argument index="0" name="label" type="String"> + </argument> + <argument index="1" name="options" type="Dictionary" default="{ + +}"> + </argument> + <description> + Returns a new [WebRTCDataChannel] (or [code]null[/code] on failure) with given [code]label[/code] and optionally configured via the [code]options[/code] dictionary. This method can only be called when the connection is in state [constant STATE_NEW]. + There are two ways to create a working data channel: either call [method create_data_channel] on only one of the peer and listen to [signal data_channel_received] on the other, or call [method create_data_channel] on both peers, with the same values, and the [code]negotiated[/code] option set to [code]true[/code]. + Valid [code]options[/code] are: + [codeblock] + { + "negotiated": true, # When set to true (default off), means the channel is negotiated out of band. "id" must be set too. data_channel_received will not be called. + "id": 1, # When "negotiated" is true this value must also be set to the same value on both peer. + + # Only one of maxRetransmits and maxPacketLifeTime can be specified, not both. They make the channel unreliable (but also better at real time). + "maxRetransmits": 1, # Specify the maximum number of attempt the peer will make to retransmits packets if they are not acknowledged. + "maxPacketLifeTime": 100, # Specify the maximum amount of time before giving up retransmitions of unacknowledged packets (in milliseconds). + "ordered": true, # When in unreliable mode (i.e. either "maxRetransmits" or "maxPacketLifetime" is set), "ordered" (true by default) specify if packet ordering is to be enforced. + + "protocol": "my-custom-protocol", # A custom sub-protocol string for this channel. + } + [/codeblock] + [b]Note:[/b] You must keep a reference to channels created this way, or it will be closed. + </description> + </method> + <method name="create_offer"> + <return type="int" enum="Error"> + </return> + <description> + Creates a new SDP offer to start a WebRTC connection with a remote peer. At least one [WebRTCDataChannel] must have been created before calling this method. + If this functions returns [constant OK], [signal session_description_created] will be called when the session is ready to be sent. + </description> + </method> + <method name="get_connection_state" qualifiers="const"> + <return type="int" enum="WebRTCPeerConnection.ConnectionState"> + </return> + <description> + Returns the connection state. See [enum ConnectionState]. + </description> + </method> + <method name="initialize"> + <return type="int" enum="Error"> + </return> + <argument index="0" name="configuration" type="Dictionary" default="{ + +}"> + </argument> + <description> + Re-initialize this peer connection, closing any previously active connection, and going back to state [constant STATE_NEW]. A dictionary of [code]options[/code] can be passed to configure the peer connection. + Valid [code]options[/code] are: + [codeblock] + { + "iceServers": [ + { + "urls": [ "stun:stun.example.com:3478" ], # One or more STUN servers. + }, + { + "urls": [ "turn:turn.example.com:3478" ], # One or more TURN servers. + "username": "a_username", # Optional username for the TURN server. + "credentials": "a_password", # Optional password for the TURN server. + } + ] + } + [/codeblock] + </description> + </method> + <method name="poll"> + <return type="int" enum="Error"> + </return> + <description> + Call this method frequently (e.g. in [method Node._process] or [method Node._physics_process]) to properly receive signals. + </description> + </method> + <method name="set_local_description"> + <return type="int" enum="Error"> + </return> + <argument index="0" name="type" type="String"> + </argument> + <argument index="1" name="sdp" type="String"> + </argument> + <description> + Sets the SDP description of the local peer. This should be called in response to [signal session_description_created]. + If [code]type[/code] is [code]answer[/code] the peer will start emitting [signal ice_candidate_created]. + </description> + </method> + <method name="set_remote_description"> + <return type="int" enum="Error"> + </return> + <argument index="0" name="type" type="String"> + </argument> + <argument index="1" name="sdp" type="String"> + </argument> + <description> + Sets the SDP description of the remote peer. This should be called with the values generated by a remote peer and received over the signaling server. + If [code]type[/code] is [code]offer[/code] the peer will emit [signal session_description_created] with the appropriate answer. + If [code]type[/code] is [code]answer[/code] the peer will start emitting [signal ice_candidate_created]. + </description> + </method> + </methods> + <signals> + <signal name="data_channel_received"> + <argument index="0" name="channel" type="Object"> + </argument> + <description> + Emitted when a new in-band channel is received, i.e. when the channel was created with [code]negotiated: false[/code] (default). + The object will be an instance of [WebRTCDataChannel]. You must keep a reference of it or it will be closed automatically. See [method create_data_channel] + </description> + </signal> + <signal name="ice_candidate_created"> + <argument index="0" name="media" type="String"> + </argument> + <argument index="1" name="index" type="int"> + </argument> + <argument index="2" name="name" type="String"> + </argument> + <description> + Emitted when a new ICE candidate has been created. The three parameters are meant to be passed to the remote peer over the signaling server. + </description> + </signal> + <signal name="session_description_created"> + <argument index="0" name="type" type="String"> + </argument> + <argument index="1" name="sdp" type="String"> + </argument> + <description> + Emitted after a successful call to [method create_offer] or [method set_remote_description] (when it generates an answer). The parameters are meant to be passed to [method set_local_description] on this object, and sent to the remote peer over the signaling server. + </description> + </signal> + </signals> + <constants> + <constant name="STATE_NEW" value="0" enum="ConnectionState"> + The connection is new, data channels and an offer can be created in this state. + </constant> + <constant name="STATE_CONNECTING" value="1" enum="ConnectionState"> + The peer is connecting, ICE is in progress, none of the transports has failed. + </constant> + <constant name="STATE_CONNECTED" value="2" enum="ConnectionState"> + The peer is connected, all ICE transports are connected. + </constant> + <constant name="STATE_DISCONNECTED" value="3" enum="ConnectionState"> + At least one ICE transport is disconnected. + </constant> + <constant name="STATE_FAILED" value="4" enum="ConnectionState"> + One or more of the ICE transports failed. + </constant> + <constant name="STATE_CLOSED" value="5" enum="ConnectionState"> + The peer connection is closed (after calling [method close] for example). + </constant> + </constants> +</class> diff --git a/modules/webrtc/register_types.cpp b/modules/webrtc/register_types.cpp new file mode 100644 index 0000000000..6f97842064 --- /dev/null +++ b/modules/webrtc/register_types.cpp @@ -0,0 +1,68 @@ +/*************************************************************************/ +/* register_types.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#include "register_types.h" +#include "core/project_settings.h" +#include "webrtc_data_channel.h" +#include "webrtc_peer_connection.h" + +#ifdef JAVASCRIPT_ENABLED +#include "emscripten.h" +#include "webrtc_peer_connection_js.h" +#endif +#ifdef WEBRTC_GDNATIVE_ENABLED +#include "webrtc_data_channel_gdnative.h" +#include "webrtc_peer_connection_gdnative.h" +#endif +#include "webrtc_multiplayer.h" + +void register_webrtc_types() { +#define _SET_HINT(NAME, _VAL_, _MAX_) \ + GLOBAL_DEF(NAME, _VAL_); \ + ProjectSettings::get_singleton()->set_custom_property_info(NAME, PropertyInfo(Variant::INT, NAME, PROPERTY_HINT_RANGE, "2," #_MAX_ ",1,or_greater")); + + _SET_HINT(WRTC_IN_BUF, 64, 4096); + +#ifdef JAVASCRIPT_ENABLED + WebRTCPeerConnectionJS::make_default(); +#elif defined(WEBRTC_GDNATIVE_ENABLED) + WebRTCPeerConnectionGDNative::make_default(); +#endif + + ClassDB::register_custom_instance_class<WebRTCPeerConnection>(); +#ifdef WEBRTC_GDNATIVE_ENABLED + ClassDB::register_class<WebRTCPeerConnectionGDNative>(); + ClassDB::register_class<WebRTCDataChannelGDNative>(); +#endif + ClassDB::register_virtual_class<WebRTCDataChannel>(); + ClassDB::register_class<WebRTCMultiplayer>(); +} + +void unregister_webrtc_types() {} diff --git a/modules/webrtc/register_types.h b/modules/webrtc/register_types.h new file mode 100644 index 0000000000..4923547a95 --- /dev/null +++ b/modules/webrtc/register_types.h @@ -0,0 +1,32 @@ +/*************************************************************************/ +/* register_types.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +void register_webrtc_types(); +void unregister_webrtc_types(); diff --git a/modules/webrtc/webrtc_data_channel.cpp b/modules/webrtc/webrtc_data_channel.cpp new file mode 100644 index 0000000000..7b3843410a --- /dev/null +++ b/modules/webrtc/webrtc_data_channel.cpp @@ -0,0 +1,66 @@ +/*************************************************************************/ +/* webrtc_data_channel.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#include "webrtc_data_channel.h" +#include "core/project_settings.h" + +void WebRTCDataChannel::_bind_methods() { + ClassDB::bind_method(D_METHOD("poll"), &WebRTCDataChannel::poll); + ClassDB::bind_method(D_METHOD("close"), &WebRTCDataChannel::close); + + ClassDB::bind_method(D_METHOD("was_string_packet"), &WebRTCDataChannel::was_string_packet); + ClassDB::bind_method(D_METHOD("set_write_mode", "write_mode"), &WebRTCDataChannel::set_write_mode); + ClassDB::bind_method(D_METHOD("get_write_mode"), &WebRTCDataChannel::get_write_mode); + ClassDB::bind_method(D_METHOD("get_ready_state"), &WebRTCDataChannel::get_ready_state); + ClassDB::bind_method(D_METHOD("get_label"), &WebRTCDataChannel::get_label); + ClassDB::bind_method(D_METHOD("is_ordered"), &WebRTCDataChannel::is_ordered); + ClassDB::bind_method(D_METHOD("get_id"), &WebRTCDataChannel::get_id); + ClassDB::bind_method(D_METHOD("get_max_packet_life_time"), &WebRTCDataChannel::get_max_packet_life_time); + ClassDB::bind_method(D_METHOD("get_max_retransmits"), &WebRTCDataChannel::get_max_retransmits); + ClassDB::bind_method(D_METHOD("get_protocol"), &WebRTCDataChannel::get_protocol); + ClassDB::bind_method(D_METHOD("is_negotiated"), &WebRTCDataChannel::is_negotiated); + + ADD_PROPERTY(PropertyInfo(Variant::INT, "write_mode", PROPERTY_HINT_ENUM), "set_write_mode", "get_write_mode"); + + BIND_ENUM_CONSTANT(WRITE_MODE_TEXT); + BIND_ENUM_CONSTANT(WRITE_MODE_BINARY); + + BIND_ENUM_CONSTANT(STATE_CONNECTING); + BIND_ENUM_CONSTANT(STATE_OPEN); + BIND_ENUM_CONSTANT(STATE_CLOSING); + BIND_ENUM_CONSTANT(STATE_CLOSED); +} + +WebRTCDataChannel::WebRTCDataChannel() { + _in_buffer_shift = nearest_shift((int)GLOBAL_GET(WRTC_IN_BUF) - 1) + 10; +} + +WebRTCDataChannel::~WebRTCDataChannel() { +} diff --git a/modules/webrtc/webrtc_data_channel.h b/modules/webrtc/webrtc_data_channel.h new file mode 100644 index 0000000000..7e2c08d9d7 --- /dev/null +++ b/modules/webrtc/webrtc_data_channel.h @@ -0,0 +1,89 @@ +/*************************************************************************/ +/* webrtc_data_channel.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifndef WEBRTC_DATA_CHANNEL_H +#define WEBRTC_DATA_CHANNEL_H + +#include "core/io/packet_peer.h" + +#define WRTC_IN_BUF "network/limits/webrtc/max_channel_in_buffer_kb" + +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: + unsigned int _in_buffer_shift; + + 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 diff --git a/modules/webrtc/webrtc_data_channel_gdnative.cpp b/modules/webrtc/webrtc_data_channel_gdnative.cpp new file mode 100644 index 0000000000..6362634626 --- /dev/null +++ b/modules/webrtc/webrtc_data_channel_gdnative.cpp @@ -0,0 +1,137 @@ +/*************************************************************************/ +/* webrtc_data_channel_gdnative.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifdef WEBRTC_GDNATIVE_ENABLED + +#include "webrtc_data_channel_gdnative.h" +#include "core/io/resource_loader.h" +#include "modules/gdnative/nativescript/nativescript.h" + +void WebRTCDataChannelGDNative::_bind_methods() { + ADD_PROPERTY_DEFAULT("write_mode", WRITE_MODE_BINARY); +} + +WebRTCDataChannelGDNative::WebRTCDataChannelGDNative() { + interface = NULL; +} + +WebRTCDataChannelGDNative::~WebRTCDataChannelGDNative() { +} + +Error WebRTCDataChannelGDNative::poll() { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->poll(interface->data); +} + +void WebRTCDataChannelGDNative::close() { + ERR_FAIL_COND(interface == NULL); + interface->close(interface->data); +} + +void WebRTCDataChannelGDNative::set_write_mode(WriteMode p_mode) { + ERR_FAIL_COND(interface == NULL); + interface->set_write_mode(interface->data, p_mode); +} + +WebRTCDataChannel::WriteMode WebRTCDataChannelGDNative::get_write_mode() const { + ERR_FAIL_COND_V(interface == NULL, WRITE_MODE_BINARY); + return (WriteMode)interface->get_write_mode(interface->data); +} + +bool WebRTCDataChannelGDNative::was_string_packet() const { + ERR_FAIL_COND_V(interface == NULL, false); + return interface->was_string_packet(interface->data); +} + +WebRTCDataChannel::ChannelState WebRTCDataChannelGDNative::get_ready_state() const { + ERR_FAIL_COND_V(interface == NULL, STATE_CLOSED); + return (ChannelState)interface->get_ready_state(interface->data); +} + +String WebRTCDataChannelGDNative::get_label() const { + ERR_FAIL_COND_V(interface == NULL, ""); + return String(interface->get_label(interface->data)); +} + +bool WebRTCDataChannelGDNative::is_ordered() const { + ERR_FAIL_COND_V(interface == NULL, false); + return interface->is_ordered(interface->data); +} + +int WebRTCDataChannelGDNative::get_id() const { + ERR_FAIL_COND_V(interface == NULL, -1); + return interface->get_id(interface->data); +} + +int WebRTCDataChannelGDNative::get_max_packet_life_time() const { + ERR_FAIL_COND_V(interface == NULL, -1); + return interface->get_max_packet_life_time(interface->data); +} + +int WebRTCDataChannelGDNative::get_max_retransmits() const { + ERR_FAIL_COND_V(interface == NULL, -1); + return interface->get_max_retransmits(interface->data); +} + +String WebRTCDataChannelGDNative::get_protocol() const { + ERR_FAIL_COND_V(interface == NULL, ""); + return String(interface->get_protocol(interface->data)); +} + +bool WebRTCDataChannelGDNative::is_negotiated() const { + ERR_FAIL_COND_V(interface == NULL, false); + return interface->is_negotiated(interface->data); +} + +Error WebRTCDataChannelGDNative::get_packet(const uint8_t **r_buffer, int &r_buffer_size) { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->get_packet(interface->data, r_buffer, &r_buffer_size); +} + +Error WebRTCDataChannelGDNative::put_packet(const uint8_t *p_buffer, int p_buffer_size) { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->put_packet(interface->data, p_buffer, p_buffer_size); +} + +int WebRTCDataChannelGDNative::get_max_packet_size() const { + ERR_FAIL_COND_V(interface == NULL, 0); + return interface->get_max_packet_size(interface->data); +} + +int WebRTCDataChannelGDNative::get_available_packet_count() const { + ERR_FAIL_COND_V(interface == NULL, 0); + return interface->get_available_packet_count(interface->data); +} + +void WebRTCDataChannelGDNative::set_native_webrtc_data_channel(const godot_net_webrtc_data_channel *p_impl) { + interface = p_impl; +} + +#endif // WEBRTC_GDNATIVE_ENABLED diff --git a/modules/webrtc/webrtc_data_channel_gdnative.h b/modules/webrtc/webrtc_data_channel_gdnative.h new file mode 100644 index 0000000000..3685f86353 --- /dev/null +++ b/modules/webrtc/webrtc_data_channel_gdnative.h @@ -0,0 +1,80 @@ +/*************************************************************************/ +/* webrtc_data_channel_gdnative.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifdef WEBRTC_GDNATIVE_ENABLED + +#ifndef WEBRTC_DATA_CHANNEL_GDNATIVE_H +#define WEBRTC_DATA_CHANNEL_GDNATIVE_H + +#include "modules/gdnative/include/net/godot_net.h" +#include "webrtc_data_channel.h" + +class WebRTCDataChannelGDNative : public WebRTCDataChannel { + GDCLASS(WebRTCDataChannelGDNative, WebRTCDataChannel); + +protected: + static void _bind_methods(); + +private: + const godot_net_webrtc_data_channel *interface; + +public: + void set_native_webrtc_data_channel(const godot_net_webrtc_data_channel *p_impl); + + 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; + + WebRTCDataChannelGDNative(); + ~WebRTCDataChannelGDNative(); +}; + +#endif // WEBRTC_DATA_CHANNEL_GDNATIVE_H + +#endif // WEBRTC_GDNATIVE_ENABLED diff --git a/modules/webrtc/webrtc_data_channel_js.cpp b/modules/webrtc/webrtc_data_channel_js.cpp new file mode 100644 index 0000000000..2edd212a50 --- /dev/null +++ b/modules/webrtc/webrtc_data_channel_js.cpp @@ -0,0 +1,365 @@ +/*************************************************************************/ +/* webrtc_data_channel_js.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifdef JAVASCRIPT_ENABLED + +#include "webrtc_data_channel_js.h" +#include "emscripten.h" + +extern "C" { +EMSCRIPTEN_KEEPALIVE void _emrtc_on_ch_error(void *obj) { + WebRTCDataChannelJS *peer = static_cast<WebRTCDataChannelJS *>(obj); + peer->_on_error(); +} + +EMSCRIPTEN_KEEPALIVE void _emrtc_on_ch_open(void *obj) { + WebRTCDataChannelJS *peer = static_cast<WebRTCDataChannelJS *>(obj); + peer->_on_open(); +} + +EMSCRIPTEN_KEEPALIVE void _emrtc_on_ch_close(void *obj) { + WebRTCDataChannelJS *peer = static_cast<WebRTCDataChannelJS *>(obj); + peer->_on_close(); +} + +EMSCRIPTEN_KEEPALIVE void _emrtc_on_ch_message(void *obj, uint8_t *p_data, uint32_t p_size, bool p_is_string) { + WebRTCDataChannelJS *peer = static_cast<WebRTCDataChannelJS *>(obj); + peer->_on_message(p_data, p_size, p_is_string); +} +} + +void WebRTCDataChannelJS::_on_open() { + in_buffer.resize(_in_buffer_shift); +} + +void WebRTCDataChannelJS::_on_close() { + close(); +} + +void WebRTCDataChannelJS::_on_error() { + close(); +} + +void WebRTCDataChannelJS::_on_message(uint8_t *p_data, uint32_t p_size, bool p_is_string) { + + ERR_FAIL_COND_MSG(in_buffer.space_left() < (int)(p_size + 5), "Buffer full! Dropping data."); + + uint8_t is_string = p_is_string ? 1 : 0; + in_buffer.write((uint8_t *)&p_size, 4); + in_buffer.write((uint8_t *)&is_string, 1); + in_buffer.write(p_data, p_size); + queue_count++; +} + +void WebRTCDataChannelJS::close() { + in_buffer.resize(0); + queue_count = 0; + _was_string = false; + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + if (!dict) return; + var channel = dict["channel"]; + channel.onopen = null; + channel.onclose = null; + channel.onerror = null; + channel.onmessage = null; + channel.close(); + }, _js_id); + /* clang-format on */ +} + +Error WebRTCDataChannelJS::poll() { + return OK; +} + +WebRTCDataChannelJS::ChannelState WebRTCDataChannelJS::get_ready_state() const { + /* clang-format off */ + return (ChannelState) EM_ASM_INT({ + var dict = Module.IDHandler.get($0); + if (!dict) return 3; // CLOSED + var channel = dict["channel"]; + switch(channel.readyState) { + case "connecting": + return 0; + case "open": + return 1; + case "closing": + return 2; + case "closed": + return 3; + } + return 3; // CLOSED + }, _js_id); + /* clang-format on */ +} + +int WebRTCDataChannelJS::get_available_packet_count() const { + return queue_count; +} + +Error WebRTCDataChannelJS::get_packet(const uint8_t **r_buffer, int &r_buffer_size) { + ERR_FAIL_COND_V(get_ready_state() != STATE_OPEN, ERR_UNCONFIGURED); + + if (queue_count == 0) + return ERR_UNAVAILABLE; + + uint32_t to_read = 0; + uint32_t left = 0; + uint8_t is_string = 0; + r_buffer_size = 0; + + in_buffer.read((uint8_t *)&to_read, 4); + --queue_count; + left = in_buffer.data_left(); + + if (left < to_read + 1) { + in_buffer.advance_read(left); + return FAILED; + } + + in_buffer.read(&is_string, 1); + _was_string = is_string == 1; + in_buffer.read(packet_buffer, to_read); + *r_buffer = packet_buffer; + r_buffer_size = to_read; + + return OK; +} + +Error WebRTCDataChannelJS::put_packet(const uint8_t *p_buffer, int p_buffer_size) { + ERR_FAIL_COND_V(get_ready_state() != STATE_OPEN, ERR_UNCONFIGURED); + + int is_bin = _write_mode == WebRTCDataChannel::WRITE_MODE_BINARY ? 1 : 0; + + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + var channel = dict["channel"]; + var bytes_array = new Uint8Array($2); + var i = 0; + + for(i=0; i<$2; i++) { + bytes_array[i] = getValue($1+i, 'i8'); + } + + if ($3) { + channel.send(bytes_array.buffer); + } else { + var string = new TextDecoder("utf-8").decode(bytes_array); + channel.send(string); + } + }, _js_id, p_buffer, p_buffer_size, is_bin); + /* clang-format on */ + + return OK; +} + +int WebRTCDataChannelJS::get_max_packet_size() const { + return 1200; +} + +void WebRTCDataChannelJS::set_write_mode(WriteMode p_mode) { + _write_mode = p_mode; +} + +WebRTCDataChannel::WriteMode WebRTCDataChannelJS::get_write_mode() const { + return _write_mode; +} + +bool WebRTCDataChannelJS::was_string_packet() const { + return _was_string; +} + +String WebRTCDataChannelJS::get_label() const { + return _label; +} + +/* clang-format off */ +#define _JS_GET(PROP, DEF) \ +EM_ASM_INT({ \ + var dict = Module.IDHandler.get($0); \ + if (!dict || !dict["channel"]) { \ + return DEF; \ + } \ + var out = dict["channel"].PROP; \ + return out === null ? DEF : out; \ +}, _js_id) +/* clang-format on */ + +bool WebRTCDataChannelJS::is_ordered() const { + return _JS_GET(ordered, true); +} + +int WebRTCDataChannelJS::get_id() const { + return _JS_GET(id, 65535); +} + +int WebRTCDataChannelJS::get_max_packet_life_time() const { + // Can't use macro, webkit workaround. + /* clang-format off */ + return EM_ASM_INT({ + var dict = Module.IDHandler.get($0); + if (!dict || !dict["channel"]) { + return 65535; + } + if (dict["channel"].maxRetransmitTime !== undefined) { + // Guess someone didn't appreciate the standardization process. + return dict["channel"].maxRetransmitTime; + } + var out = dict["channel"].maxPacketLifeTime; + return out === null ? 65535 : out; + }, _js_id); + /* clang-format on */ +} + +int WebRTCDataChannelJS::get_max_retransmits() const { + return _JS_GET(maxRetransmits, 65535); +} + +String WebRTCDataChannelJS::get_protocol() const { + return _protocol; +} + +bool WebRTCDataChannelJS::is_negotiated() const { + return _JS_GET(negotiated, false); +} + +WebRTCDataChannelJS::WebRTCDataChannelJS() { + queue_count = 0; + _was_string = false; + _write_mode = WRITE_MODE_BINARY; + _js_id = 0; +} + +WebRTCDataChannelJS::WebRTCDataChannelJS(int js_id) { + queue_count = 0; + _was_string = false; + _write_mode = WRITE_MODE_BINARY; + _js_id = js_id; + + /* clang-format off */ + EM_ASM({ + var c_ptr = $0; + var dict = Module.IDHandler.get($1); + if (!dict) return; + var channel = dict["channel"]; + dict["ptr"] = c_ptr; + + channel.binaryType = "arraybuffer"; + channel.onopen = function (evt) { + ccall("_emrtc_on_ch_open", + "void", + ["number"], + [c_ptr] + ); + }; + channel.onclose = function (evt) { + ccall("_emrtc_on_ch_close", + "void", + ["number"], + [c_ptr] + ); + }; + channel.onerror = function (evt) { + ccall("_emrtc_on_ch_error", + "void", + ["number"], + [c_ptr] + ); + }; + channel.onmessage = function(event) { + var buffer; + var is_string = 0; + if (event.data instanceof ArrayBuffer) { + buffer = new Uint8Array(event.data); + } else if (event.data instanceof Blob) { + console.error("Blob type not supported"); + return; + } else if (typeof event.data === "string") { + is_string = 1; + var enc = new TextEncoder("utf-8"); + buffer = new Uint8Array(enc.encode(event.data)); + } else { + console.error("Unknown message type"); + return; + } + var len = buffer.length*buffer.BYTES_PER_ELEMENT; + var out = Module._malloc(len); + Module.HEAPU8.set(buffer, out); + ccall("_emrtc_on_ch_message", + "void", + ["number", "number", "number", "number"], + [c_ptr, out, len, is_string] + ); + Module._free(out); + } + + }, this, js_id); + // Parse label + char *str; + str = (char *)EM_ASM_INT({ + var dict = Module.IDHandler.get($0); + if (!dict || !dict["channel"]) return 0; + var str = dict["channel"].label; + var len = lengthBytesUTF8(str)+1; + var ptr = _malloc(str); + stringToUTF8(str, ptr, len+1); + return ptr; + }, js_id); + if(str != NULL) { + _label.parse_utf8(str); + EM_ASM({ _free($0) }, str); + } + str = (char *)EM_ASM_INT({ + var dict = Module.IDHandler.get($0); + if (!dict || !dict["channel"]) return 0; + var str = dict["channel"].protocol; + var len = lengthBytesUTF8(str)+1; + var ptr = _malloc(str); + stringToUTF8(str, ptr, len+1); + return ptr; + }, js_id); + if(str != NULL) { + _protocol.parse_utf8(str); + EM_ASM({ _free($0) }, str); + } + /* clang-format on */ +} + +WebRTCDataChannelJS::~WebRTCDataChannelJS() { + close(); + /* clang-format off */ + EM_ASM({ + Module.IDHandler.remove($0); + }, _js_id); + /* clang-format on */ +}; +#endif diff --git a/modules/webrtc/webrtc_data_channel_js.h b/modules/webrtc/webrtc_data_channel_js.h new file mode 100644 index 0000000000..b87f8e9326 --- /dev/null +++ b/modules/webrtc/webrtc_data_channel_js.h @@ -0,0 +1,93 @@ +/*************************************************************************/ +/* webrtc_data_channel_js.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#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 diff --git a/modules/webrtc/webrtc_multiplayer.cpp b/modules/webrtc/webrtc_multiplayer.cpp new file mode 100644 index 0000000000..a759b17b83 --- /dev/null +++ b/modules/webrtc/webrtc_multiplayer.cpp @@ -0,0 +1,382 @@ +/*************************************************************************/ +/* webrtc_multiplayer.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#include "webrtc_multiplayer.h" + +#include "core/io/marshalls.h" +#include "core/os/os.h" + +void WebRTCMultiplayer::_bind_methods() { + ClassDB::bind_method(D_METHOD("initialize", "peer_id", "server_compatibility"), &WebRTCMultiplayer::initialize, DEFVAL(false)); + ClassDB::bind_method(D_METHOD("add_peer", "peer", "peer_id", "unreliable_lifetime"), &WebRTCMultiplayer::add_peer, DEFVAL(1)); + ClassDB::bind_method(D_METHOD("remove_peer", "peer_id"), &WebRTCMultiplayer::remove_peer); + ClassDB::bind_method(D_METHOD("has_peer", "peer_id"), &WebRTCMultiplayer::has_peer); + ClassDB::bind_method(D_METHOD("get_peer", "peer_id"), &WebRTCMultiplayer::get_peer); + ClassDB::bind_method(D_METHOD("get_peers"), &WebRTCMultiplayer::get_peers); + ClassDB::bind_method(D_METHOD("close"), &WebRTCMultiplayer::close); +} + +void WebRTCMultiplayer::set_transfer_mode(TransferMode p_mode) { + transfer_mode = p_mode; +} + +NetworkedMultiplayerPeer::TransferMode WebRTCMultiplayer::get_transfer_mode() const { + return transfer_mode; +} + +void WebRTCMultiplayer::set_target_peer(int p_peer_id) { + target_peer = p_peer_id; +} + +/* Returns the ID of the NetworkedMultiplayerPeer who sent the most recent packet: */ +int WebRTCMultiplayer::get_packet_peer() const { + return next_packet_peer; +} + +bool WebRTCMultiplayer::is_server() const { + return unique_id == TARGET_PEER_SERVER; +} + +void WebRTCMultiplayer::poll() { + if (peer_map.size() == 0) + return; + + List<int> remove; + List<int> add; + for (Map<int, Ref<ConnectedPeer> >::Element *E = peer_map.front(); E; E = E->next()) { + Ref<ConnectedPeer> peer = E->get(); + peer->connection->poll(); + // Check peer state + switch (peer->connection->get_connection_state()) { + case WebRTCPeerConnection::STATE_NEW: + case WebRTCPeerConnection::STATE_CONNECTING: + // Go to next peer, not ready yet. + continue; + case WebRTCPeerConnection::STATE_CONNECTED: + // Good to go, go ahead and check channel state. + break; + default: + // Peer is closed or in error state. Got to next peer. + remove.push_back(E->key()); + continue; + } + // Check channels state + int ready = 0; + for (List<Ref<WebRTCDataChannel> >::Element *C = peer->channels.front(); C && C->get().is_valid(); C = C->next()) { + Ref<WebRTCDataChannel> ch = C->get(); + switch (ch->get_ready_state()) { + case WebRTCDataChannel::STATE_CONNECTING: + continue; + case WebRTCDataChannel::STATE_OPEN: + ready++; + continue; + default: + // Channel was closed or in error state, remove peer id. + remove.push_back(E->key()); + } + // We got a closed channel break out, the peer will be removed. + break; + } + // This peer has newly connected, and all channels are now open. + if (ready == peer->channels.size() && !peer->connected) { + peer->connected = true; + add.push_back(E->key()); + } + } + // Remove disconnected peers + for (List<int>::Element *E = remove.front(); E; E = E->next()) { + remove_peer(E->get()); + if (next_packet_peer == E->get()) + next_packet_peer = 0; + } + // Signal newly connected peers + for (List<int>::Element *E = add.front(); E; E = E->next()) { + // Already connected to server: simply notify new peer. + // NOTE: Mesh is always connected. + if (connection_status == CONNECTION_CONNECTED) + emit_signal("peer_connected", E->get()); + + // Server emulation mode suppresses peer_conencted until server connects. + if (server_compat && E->get() == TARGET_PEER_SERVER) { + // Server connected. + connection_status = CONNECTION_CONNECTED; + emit_signal("peer_connected", TARGET_PEER_SERVER); + emit_signal("connection_succeeded"); + // Notify of all previously connected peers + for (Map<int, Ref<ConnectedPeer> >::Element *F = peer_map.front(); F; F = F->next()) { + if (F->key() != 1 && F->get()->connected) + emit_signal("peer_connected", F->key()); + } + break; // Because we already notified of all newly added peers. + } + } + // Fetch next packet + if (next_packet_peer == 0) + _find_next_peer(); +} + +void WebRTCMultiplayer::_find_next_peer() { + Map<int, Ref<ConnectedPeer> >::Element *E = peer_map.find(next_packet_peer); + if (E) E = E->next(); + // After last. + while (E) { + for (List<Ref<WebRTCDataChannel> >::Element *F = E->get()->channels.front(); F; F = F->next()) { + if (F->get()->get_available_packet_count()) { + next_packet_peer = E->key(); + return; + } + } + E = E->next(); + } + E = peer_map.front(); + // Before last + while (E) { + for (List<Ref<WebRTCDataChannel> >::Element *F = E->get()->channels.front(); F; F = F->next()) { + if (F->get()->get_available_packet_count()) { + next_packet_peer = E->key(); + return; + } + } + if (E->key() == (int)next_packet_peer) + break; + E = E->next(); + } + // No packet found + next_packet_peer = 0; +} + +void WebRTCMultiplayer::set_refuse_new_connections(bool p_enable) { + refuse_connections = p_enable; +} + +bool WebRTCMultiplayer::is_refusing_new_connections() const { + return refuse_connections; +} + +NetworkedMultiplayerPeer::ConnectionStatus WebRTCMultiplayer::get_connection_status() const { + return connection_status; +} + +Error WebRTCMultiplayer::initialize(int p_self_id, bool p_server_compat) { + ERR_FAIL_COND_V(p_self_id < 0 || p_self_id > ~(1 << 31), ERR_INVALID_PARAMETER); + unique_id = p_self_id; + server_compat = p_server_compat; + + // Mesh and server are always connected + if (!server_compat || p_self_id == 1) + connection_status = CONNECTION_CONNECTED; + else + connection_status = CONNECTION_CONNECTING; + return OK; +} + +int WebRTCMultiplayer::get_unique_id() const { + ERR_FAIL_COND_V(connection_status == CONNECTION_DISCONNECTED, 1); + return unique_id; +} + +void WebRTCMultiplayer::_peer_to_dict(Ref<ConnectedPeer> p_connected_peer, Dictionary &r_dict) { + Array channels; + for (List<Ref<WebRTCDataChannel> >::Element *F = p_connected_peer->channels.front(); F; F = F->next()) { + channels.push_back(F->get()); + } + r_dict["connection"] = p_connected_peer->connection; + r_dict["connected"] = p_connected_peer->connected; + r_dict["channels"] = channels; +} + +bool WebRTCMultiplayer::has_peer(int p_peer_id) { + return peer_map.has(p_peer_id); +} + +Dictionary WebRTCMultiplayer::get_peer(int p_peer_id) { + ERR_FAIL_COND_V(!peer_map.has(p_peer_id), Dictionary()); + Dictionary out; + _peer_to_dict(peer_map[p_peer_id], out); + return out; +} + +Dictionary WebRTCMultiplayer::get_peers() { + Dictionary out; + for (Map<int, Ref<ConnectedPeer> >::Element *E = peer_map.front(); E; E = E->next()) { + Dictionary d; + _peer_to_dict(E->get(), d); + out[E->key()] = d; + } + return out; +} + +Error WebRTCMultiplayer::add_peer(Ref<WebRTCPeerConnection> p_peer, int p_peer_id, int p_unreliable_lifetime) { + ERR_FAIL_COND_V(p_peer_id < 0 || p_peer_id > ~(1 << 31), ERR_INVALID_PARAMETER); + ERR_FAIL_COND_V(p_unreliable_lifetime < 0, ERR_INVALID_PARAMETER); + ERR_FAIL_COND_V(refuse_connections, ERR_UNAUTHORIZED); + // Peer must be valid, and in new state (to create data channels) + ERR_FAIL_COND_V(!p_peer.is_valid(), ERR_INVALID_PARAMETER); + ERR_FAIL_COND_V(p_peer->get_connection_state() != WebRTCPeerConnection::STATE_NEW, ERR_INVALID_PARAMETER); + + Ref<ConnectedPeer> peer = memnew(ConnectedPeer); + peer->connection = p_peer; + + // Initialize data channels + Dictionary cfg; + cfg["negotiated"] = true; + cfg["ordered"] = true; + + cfg["id"] = 1; + peer->channels[CH_RELIABLE] = p_peer->create_data_channel("reliable", cfg); + ERR_FAIL_COND_V(!peer->channels[CH_RELIABLE].is_valid(), FAILED); + + cfg["id"] = 2; + cfg["maxPacketLifetime"] = p_unreliable_lifetime; + peer->channels[CH_ORDERED] = p_peer->create_data_channel("ordered", cfg); + ERR_FAIL_COND_V(!peer->channels[CH_ORDERED].is_valid(), FAILED); + + cfg["id"] = 3; + cfg["ordered"] = false; + peer->channels[CH_UNRELIABLE] = p_peer->create_data_channel("unreliable", cfg); + ERR_FAIL_COND_V(!peer->channels[CH_UNRELIABLE].is_valid(), FAILED); + + peer_map[p_peer_id] = peer; // add the new peer connection to the peer_map + + return OK; +} + +void WebRTCMultiplayer::remove_peer(int p_peer_id) { + ERR_FAIL_COND(!peer_map.has(p_peer_id)); + Ref<ConnectedPeer> peer = peer_map[p_peer_id]; + peer_map.erase(p_peer_id); + if (peer->connected) { + peer->connected = false; + emit_signal("peer_disconnected", p_peer_id); + if (server_compat && p_peer_id == TARGET_PEER_SERVER) { + emit_signal("server_disconnected"); + connection_status = CONNECTION_DISCONNECTED; + } + } +} + +Error WebRTCMultiplayer::get_packet(const uint8_t **r_buffer, int &r_buffer_size) { + // Peer not available + if (next_packet_peer == 0 || !peer_map.has(next_packet_peer)) { + _find_next_peer(); + ERR_FAIL_V(ERR_UNAVAILABLE); + } + for (List<Ref<WebRTCDataChannel> >::Element *E = peer_map[next_packet_peer]->channels.front(); E; E = E->next()) { + if (E->get()->get_available_packet_count()) { + Error err = E->get()->get_packet(r_buffer, r_buffer_size); + _find_next_peer(); + return err; + } + } + // Channels for that peer were empty. Bug? + _find_next_peer(); + ERR_FAIL_V(ERR_BUG); +} + +Error WebRTCMultiplayer::put_packet(const uint8_t *p_buffer, int p_buffer_size) { + ERR_FAIL_COND_V(connection_status == CONNECTION_DISCONNECTED, ERR_UNCONFIGURED); + + int ch = CH_RELIABLE; + switch (transfer_mode) { + case TRANSFER_MODE_RELIABLE: + ch = CH_RELIABLE; + break; + case TRANSFER_MODE_UNRELIABLE_ORDERED: + ch = CH_ORDERED; + break; + case TRANSFER_MODE_UNRELIABLE: + ch = CH_UNRELIABLE; + break; + } + + Map<int, Ref<ConnectedPeer> >::Element *E = NULL; + + if (target_peer > 0) { + + E = peer_map.find(target_peer); + ERR_FAIL_COND_V_MSG(!E, ERR_INVALID_PARAMETER, "Invalid target peer: " + itos(target_peer) + "."); + + ERR_FAIL_COND_V(E->value()->channels.size() <= ch, ERR_BUG); + ERR_FAIL_COND_V(!E->value()->channels[ch].is_valid(), ERR_BUG); + return E->value()->channels[ch]->put_packet(p_buffer, p_buffer_size); + + } else { + int exclude = -target_peer; + + for (Map<int, Ref<ConnectedPeer> >::Element *F = peer_map.front(); F; F = F->next()) { + + // Exclude packet. If target_peer == 0 then don't exclude any packets + if (target_peer != 0 && F->key() == exclude) + continue; + + ERR_CONTINUE(F->value()->channels.size() <= ch || !F->value()->channels[ch].is_valid()); + F->value()->channels[ch]->put_packet(p_buffer, p_buffer_size); + } + } + return OK; +} + +int WebRTCMultiplayer::get_available_packet_count() const { + if (next_packet_peer == 0) + return 0; // To be sure next call to get_packet works if size > 0 . + int size = 0; + for (Map<int, Ref<ConnectedPeer> >::Element *E = peer_map.front(); E; E = E->next()) { + for (List<Ref<WebRTCDataChannel> >::Element *F = E->get()->channels.front(); F; F = F->next()) { + size += F->get()->get_available_packet_count(); + } + } + return size; +} + +int WebRTCMultiplayer::get_max_packet_size() const { + return 1200; +} + +void WebRTCMultiplayer::close() { + peer_map.clear(); + unique_id = 0; + next_packet_peer = 0; + target_peer = 0; + connection_status = CONNECTION_DISCONNECTED; +} + +WebRTCMultiplayer::WebRTCMultiplayer() { + unique_id = 0; + next_packet_peer = 0; + target_peer = 0; + transfer_mode = TRANSFER_MODE_RELIABLE; + refuse_connections = false; + connection_status = CONNECTION_DISCONNECTED; + server_compat = false; +} + +WebRTCMultiplayer::~WebRTCMultiplayer() { + close(); +} diff --git a/modules/webrtc/webrtc_multiplayer.h b/modules/webrtc/webrtc_multiplayer.h new file mode 100644 index 0000000000..82bbfd4f68 --- /dev/null +++ b/modules/webrtc/webrtc_multiplayer.h @@ -0,0 +1,116 @@ +/*************************************************************************/ +/* webrtc_multiplayer.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifndef WEBRTC_MULTIPLAYER_H +#define WEBRTC_MULTIPLAYER_H + +#include "core/io/networked_multiplayer_peer.h" +#include "webrtc_peer_connection.h" + +class WebRTCMultiplayer : public NetworkedMultiplayerPeer { + + GDCLASS(WebRTCMultiplayer, NetworkedMultiplayerPeer); + +protected: + static void _bind_methods(); + +private: + enum { + CH_RELIABLE = 0, + CH_ORDERED = 1, + CH_UNRELIABLE = 2, + CH_RESERVED_MAX = 3 + }; + + class ConnectedPeer : public Reference { + + public: + Ref<WebRTCPeerConnection> connection; + List<Ref<WebRTCDataChannel> > channels; + bool connected; + + ConnectedPeer() { + connected = false; + for (int i = 0; i < CH_RESERVED_MAX; i++) + channels.push_front(Ref<WebRTCDataChannel>()); + } + }; + + uint32_t unique_id; + int target_peer; + int client_count; + bool refuse_connections; + ConnectionStatus connection_status; + TransferMode transfer_mode; + int next_packet_peer; + bool server_compat; + + Map<int, Ref<ConnectedPeer> > peer_map; + + void _peer_to_dict(Ref<ConnectedPeer> p_connected_peer, Dictionary &r_dict); + void _find_next_peer(); + +public: + WebRTCMultiplayer(); + ~WebRTCMultiplayer(); + + Error initialize(int p_self_id, bool p_server_compat = false); + Error add_peer(Ref<WebRTCPeerConnection> p_peer, int p_peer_id, int p_unreliable_lifetime = 1); + void remove_peer(int p_peer_id); + bool has_peer(int p_peer_id); + Dictionary get_peer(int p_peer_id); + Dictionary get_peers(); + void close(); + + // PacketPeer + Error get_packet(const uint8_t **r_buffer, int &r_buffer_size); ///< buffer is GONE after next get_packet + Error put_packet(const uint8_t *p_buffer, int p_buffer_size); + int get_available_packet_count() const; + int get_max_packet_size() const; + + // NetworkedMultiplayerPeer + void set_transfer_mode(TransferMode p_mode); + TransferMode get_transfer_mode() const; + void set_target_peer(int p_peer_id); + + int get_unique_id() const; + int get_packet_peer() const; + + bool is_server() const; + + void poll(); + + void set_refuse_new_connections(bool p_enable); + bool is_refusing_new_connections() const; + + ConnectionStatus get_connection_status() const; +}; + +#endif diff --git a/modules/webrtc/webrtc_peer_connection.cpp b/modules/webrtc/webrtc_peer_connection.cpp new file mode 100644 index 0000000000..69c7a51a40 --- /dev/null +++ b/modules/webrtc/webrtc_peer_connection.cpp @@ -0,0 +1,75 @@ +/*************************************************************************/ +/* webrtc_peer_connection.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#include "webrtc_peer_connection.h" + +WebRTCPeerConnection *(*WebRTCPeerConnection::_create)() = NULL; + +Ref<WebRTCPeerConnection> WebRTCPeerConnection::create_ref() { + + return create(); +} + +WebRTCPeerConnection *WebRTCPeerConnection::create() { + + if (!_create) + return NULL; + return _create(); +} + +void WebRTCPeerConnection::_bind_methods() { + ClassDB::bind_method(D_METHOD("initialize", "configuration"), &WebRTCPeerConnection::initialize, DEFVAL(Dictionary())); + ClassDB::bind_method(D_METHOD("create_data_channel", "label", "options"), &WebRTCPeerConnection::create_data_channel, DEFVAL(Dictionary())); + ClassDB::bind_method(D_METHOD("create_offer"), &WebRTCPeerConnection::create_offer); + ClassDB::bind_method(D_METHOD("set_local_description", "type", "sdp"), &WebRTCPeerConnection::set_local_description); + ClassDB::bind_method(D_METHOD("set_remote_description", "type", "sdp"), &WebRTCPeerConnection::set_remote_description); + ClassDB::bind_method(D_METHOD("add_ice_candidate", "media", "index", "name"), &WebRTCPeerConnection::add_ice_candidate); + ClassDB::bind_method(D_METHOD("poll"), &WebRTCPeerConnection::poll); + ClassDB::bind_method(D_METHOD("close"), &WebRTCPeerConnection::close); + + ClassDB::bind_method(D_METHOD("get_connection_state"), &WebRTCPeerConnection::get_connection_state); + + ADD_SIGNAL(MethodInfo("session_description_created", PropertyInfo(Variant::STRING, "type"), PropertyInfo(Variant::STRING, "sdp"))); + ADD_SIGNAL(MethodInfo("ice_candidate_created", PropertyInfo(Variant::STRING, "media"), PropertyInfo(Variant::INT, "index"), PropertyInfo(Variant::STRING, "name"))); + ADD_SIGNAL(MethodInfo("data_channel_received", PropertyInfo(Variant::OBJECT, "channel"))); + + BIND_ENUM_CONSTANT(STATE_NEW); + BIND_ENUM_CONSTANT(STATE_CONNECTING); + BIND_ENUM_CONSTANT(STATE_CONNECTED); + BIND_ENUM_CONSTANT(STATE_DISCONNECTED); + BIND_ENUM_CONSTANT(STATE_FAILED); + BIND_ENUM_CONSTANT(STATE_CLOSED); +} + +WebRTCPeerConnection::WebRTCPeerConnection() { +} + +WebRTCPeerConnection::~WebRTCPeerConnection() { +} diff --git a/modules/webrtc/webrtc_peer_connection.h b/modules/webrtc/webrtc_peer_connection.h new file mode 100644 index 0000000000..7be1390dab --- /dev/null +++ b/modules/webrtc/webrtc_peer_connection.h @@ -0,0 +1,74 @@ +/*************************************************************************/ +/* webrtc_peer_connection.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifndef WEBRTC_PEER_CONNECTION_H +#define WEBRTC_PEER_CONNECTION_H + +#include "core/io/packet_peer.h" +#include "modules/webrtc/webrtc_data_channel.h" + +class WebRTCPeerConnection : public Reference { + GDCLASS(WebRTCPeerConnection, Reference); + +public: + enum ConnectionState { + STATE_NEW, + STATE_CONNECTING, + STATE_CONNECTED, + STATE_DISCONNECTED, + STATE_FAILED, + STATE_CLOSED + }; + +protected: + static void _bind_methods(); + static WebRTCPeerConnection *(*_create)(); + +public: + virtual ConnectionState get_connection_state() const = 0; + + virtual Error initialize(Dictionary p_config = Dictionary()) = 0; + virtual Ref<WebRTCDataChannel> create_data_channel(String p_label, Dictionary p_options = Dictionary()) = 0; + virtual Error create_offer() = 0; + virtual Error set_remote_description(String type, String sdp) = 0; + virtual Error set_local_description(String type, String sdp) = 0; + virtual Error add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName) = 0; + virtual Error poll() = 0; + virtual void close() = 0; + + static Ref<WebRTCPeerConnection> create_ref(); + static WebRTCPeerConnection *create(); + + WebRTCPeerConnection(); + ~WebRTCPeerConnection(); +}; + +VARIANT_ENUM_CAST(WebRTCPeerConnection::ConnectionState); +#endif // WEBRTC_PEER_CONNECTION_H diff --git a/modules/webrtc/webrtc_peer_connection_gdnative.cpp b/modules/webrtc/webrtc_peer_connection_gdnative.cpp new file mode 100644 index 0000000000..5e9dcb5366 --- /dev/null +++ b/modules/webrtc/webrtc_peer_connection_gdnative.cpp @@ -0,0 +1,122 @@ +/*************************************************************************/ +/* webrtc_peer_connection_gdnative.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifdef WEBRTC_GDNATIVE_ENABLED + +#include "webrtc_peer_connection_gdnative.h" + +#include "core/io/resource_loader.h" +#include "modules/gdnative/nativescript/nativescript.h" +#include "webrtc_data_channel_gdnative.h" + +const godot_net_webrtc_library *WebRTCPeerConnectionGDNative::default_library = NULL; + +Error WebRTCPeerConnectionGDNative::set_default_library(const godot_net_webrtc_library *p_lib) { + if (default_library) { + const godot_net_webrtc_library *old = default_library; + default_library = NULL; + old->unregistered(); + } + default_library = p_lib; + return OK; // Maybe add version check and fail accordingly +} + +WebRTCPeerConnection *WebRTCPeerConnectionGDNative::_create() { + + WebRTCPeerConnectionGDNative *obj = memnew(WebRTCPeerConnectionGDNative); + ERR_FAIL_COND_V_MSG(!default_library, obj, "Default GDNative WebRTC implementation not defined."); + + // Call GDNative constructor + Error err = (Error)default_library->create_peer_connection(obj); + ERR_FAIL_COND_V_MSG(err != OK, obj, "GDNative default library constructor returned an error."); + + return obj; +} + +void WebRTCPeerConnectionGDNative::_bind_methods() { +} + +WebRTCPeerConnectionGDNative::WebRTCPeerConnectionGDNative() { + interface = NULL; +} + +WebRTCPeerConnectionGDNative::~WebRTCPeerConnectionGDNative() { +} + +Error WebRTCPeerConnectionGDNative::initialize(Dictionary p_config) { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->initialize(interface->data, (const godot_dictionary *)&p_config); +} + +Ref<WebRTCDataChannel> WebRTCPeerConnectionGDNative::create_data_channel(String p_label, Dictionary p_options) { + ERR_FAIL_COND_V(interface == NULL, NULL); + return (WebRTCDataChannel *)interface->create_data_channel(interface->data, p_label.utf8().get_data(), (const godot_dictionary *)&p_options); +} + +Error WebRTCPeerConnectionGDNative::create_offer() { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->create_offer(interface->data); +} + +Error WebRTCPeerConnectionGDNative::set_local_description(String p_type, String p_sdp) { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->set_local_description(interface->data, p_type.utf8().get_data(), p_sdp.utf8().get_data()); +} + +Error WebRTCPeerConnectionGDNative::set_remote_description(String p_type, String p_sdp) { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->set_remote_description(interface->data, p_type.utf8().get_data(), p_sdp.utf8().get_data()); +} + +Error WebRTCPeerConnectionGDNative::add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName) { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->add_ice_candidate(interface->data, sdpMidName.utf8().get_data(), sdpMlineIndexName, sdpName.utf8().get_data()); +} + +Error WebRTCPeerConnectionGDNative::poll() { + ERR_FAIL_COND_V(interface == NULL, ERR_UNCONFIGURED); + return (Error)interface->poll(interface->data); +} + +void WebRTCPeerConnectionGDNative::close() { + ERR_FAIL_COND(interface == NULL); + interface->close(interface->data); +} + +WebRTCPeerConnection::ConnectionState WebRTCPeerConnectionGDNative::get_connection_state() const { + ERR_FAIL_COND_V(interface == NULL, STATE_DISCONNECTED); + return (ConnectionState)interface->get_connection_state(interface->data); +} + +void WebRTCPeerConnectionGDNative::set_native_webrtc_peer_connection(const godot_net_webrtc_peer_connection *p_impl) { + interface = p_impl; +} + +#endif // WEBRTC_GDNATIVE_ENABLED diff --git a/modules/webrtc/webrtc_peer_connection_gdnative.h b/modules/webrtc/webrtc_peer_connection_gdnative.h new file mode 100644 index 0000000000..0a281c3d89 --- /dev/null +++ b/modules/webrtc/webrtc_peer_connection_gdnative.h @@ -0,0 +1,73 @@ +/*************************************************************************/ +/* webrtc_peer_connection_gdnative.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifdef WEBRTC_GDNATIVE_ENABLED + +#ifndef WEBRTC_PEER_CONNECTION_GDNATIVE_H +#define WEBRTC_PEER_CONNECTION_GDNATIVE_H + +#include "modules/gdnative/include/net/godot_net.h" +#include "webrtc_peer_connection.h" + +class WebRTCPeerConnectionGDNative : public WebRTCPeerConnection { + GDCLASS(WebRTCPeerConnectionGDNative, WebRTCPeerConnection); + +protected: + static void _bind_methods(); + static WebRTCPeerConnection *_create(); + +private: + static const godot_net_webrtc_library *default_library; + const godot_net_webrtc_peer_connection *interface; + +public: + static Error set_default_library(const godot_net_webrtc_library *p_library); + static void make_default() { WebRTCPeerConnection::_create = WebRTCPeerConnectionGDNative::_create; } + + void set_native_webrtc_peer_connection(const godot_net_webrtc_peer_connection *p_impl); + + virtual ConnectionState get_connection_state() const; + + virtual Error initialize(Dictionary p_config = Dictionary()); + virtual Ref<WebRTCDataChannel> create_data_channel(String p_label, Dictionary p_options = Dictionary()); + virtual Error create_offer(); + virtual Error set_remote_description(String type, String sdp); + virtual Error set_local_description(String type, String sdp); + virtual Error add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName); + virtual Error poll(); + virtual void close(); + + WebRTCPeerConnectionGDNative(); + ~WebRTCPeerConnectionGDNative(); +}; + +#endif // WEBRTC_PEER_CONNECTION_GDNATIVE_H + +#endif // WEBRTC_GDNATIVE_ENABLED diff --git a/modules/webrtc/webrtc_peer_connection_js.cpp b/modules/webrtc/webrtc_peer_connection_js.cpp new file mode 100644 index 0000000000..9758ab3644 --- /dev/null +++ b/modules/webrtc/webrtc_peer_connection_js.cpp @@ -0,0 +1,314 @@ +/*************************************************************************/ +/* webrtc_peer_connection_js.cpp */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifdef JAVASCRIPT_ENABLED + +#include "webrtc_peer_connection_js.h" + +#include "webrtc_data_channel_js.h" + +#include "core/io/json.h" +#include "emscripten.h" + +extern "C" { +EMSCRIPTEN_KEEPALIVE void _emrtc_on_ice_candidate(void *obj, char *p_MidName, int p_MlineIndexName, char *p_sdpName) { + WebRTCPeerConnectionJS *peer = static_cast<WebRTCPeerConnectionJS *>(obj); + peer->emit_signal("ice_candidate_created", String(p_MidName), p_MlineIndexName, String(p_sdpName)); +} + +EMSCRIPTEN_KEEPALIVE void _emrtc_session_description_created(void *obj, char *p_type, char *p_offer) { + WebRTCPeerConnectionJS *peer = static_cast<WebRTCPeerConnectionJS *>(obj); + peer->emit_signal("session_description_created", String(p_type), String(p_offer)); +} + +EMSCRIPTEN_KEEPALIVE void _emrtc_on_connection_state_changed(void *obj) { + WebRTCPeerConnectionJS *peer = static_cast<WebRTCPeerConnectionJS *>(obj); + peer->_on_connection_state_changed(); +} + +EMSCRIPTEN_KEEPALIVE void _emrtc_on_error() { + ERR_PRINT("RTCPeerConnection error!"); +} + +EMSCRIPTEN_KEEPALIVE void _emrtc_emit_channel(void *obj, int p_id) { + WebRTCPeerConnectionJS *peer = static_cast<WebRTCPeerConnectionJS *>(obj); + peer->emit_signal("data_channel_received", Ref<WebRTCDataChannelJS>(new WebRTCDataChannelJS(p_id))); +} +} + +void _emrtc_create_pc(int p_id, const Dictionary &p_config) { + String config = JSON::print(p_config); + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + var c_ptr = dict["ptr"]; + var config = JSON.parse(UTF8ToString($1)); + // Setup local connaction + var conn = null; + try { + conn = new RTCPeerConnection(config); + } catch (e) { + console.log(e); + return; + } + conn.oniceconnectionstatechange = function(event) { + if (!Module.IDHandler.get($0)) return; + ccall("_emrtc_on_connection_state_changed", "void", ["number"], [c_ptr]); + }; + conn.onicecandidate = function(event) { + if (!Module.IDHandler.get($0)) return; + if (!event.candidate) return; + + var c = event.candidate; + // should emit on ice candidate + ccall("_emrtc_on_ice_candidate", + "void", + ["number", "string", "number", "string"], + [c_ptr, c.sdpMid, c.sdpMLineIndex, c.candidate] + ); + }; + conn.ondatachannel = function (evt) { + var dict = Module.IDHandler.get($0); + if (!dict) { + return; + } + var id = Module.IDHandler.add({"channel": evt.channel, "ptr": null}); + ccall("_emrtc_emit_channel", + "void", + ["number", "number"], + [c_ptr, id] + ); + }; + dict["conn"] = conn; + }, p_id, config.utf8().get_data()); + /* clang-format on */ +} + +void WebRTCPeerConnectionJS::_on_connection_state_changed() { + /* clang-format off */ + _conn_state = (ConnectionState)EM_ASM_INT({ + var dict = Module.IDHandler.get($0); + if (!dict) return 5; // CLOSED + var conn = dict["conn"]; + switch(conn.iceConnectionState) { + case "new": + return 0; + case "checking": + return 1; + case "connected": + case "completed": + return 2; + case "disconnected": + return 3; + case "failed": + return 4; + case "closed": + return 5; + } + return 5; // CLOSED + }, _js_id); + /* clang-format on */ +} + +void WebRTCPeerConnectionJS::close() { + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + if (!dict) return; + if (dict["conn"]) { + dict["conn"].close(); + } + }, _js_id); + /* clang-format on */ + _conn_state = STATE_CLOSED; +} + +Error WebRTCPeerConnectionJS::create_offer() { + ERR_FAIL_COND_V(_conn_state != STATE_NEW, FAILED); + + _conn_state = STATE_CONNECTING; + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + var conn = dict["conn"]; + var c_ptr = dict["ptr"]; + var onError = function(error) { + console.error(error); + ccall("_emrtc_on_error", "void", [], []); + }; + var onCreated = function(offer) { + ccall("_emrtc_session_description_created", + "void", + ["number", "string", "string"], + [c_ptr, offer.type, offer.sdp] + ); + }; + conn.createOffer().then(onCreated).catch(onError); + }, _js_id); + /* clang-format on */ + return OK; +} + +Error WebRTCPeerConnectionJS::set_local_description(String type, String sdp) { + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + var conn = dict["conn"]; + var c_ptr = dict["ptr"]; + var type = UTF8ToString($1); + var sdp = UTF8ToString($2); + var onError = function(error) { + console.error(error); + ccall("_emrtc_on_error", "void", [], []); + }; + conn.setLocalDescription({ + "sdp": sdp, + "type": type + }).catch(onError); + }, _js_id, type.utf8().get_data(), sdp.utf8().get_data()); + /* clang-format on */ + return OK; +} + +Error WebRTCPeerConnectionJS::set_remote_description(String type, String sdp) { + if (type == "offer") { + ERR_FAIL_COND_V(_conn_state != STATE_NEW, FAILED); + _conn_state = STATE_CONNECTING; + } + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + var conn = dict["conn"]; + var c_ptr = dict["ptr"]; + var type = UTF8ToString($1); + var sdp = UTF8ToString($2); + + var onError = function(error) { + console.error(error); + ccall("_emrtc_on_error", "void", [], []); + }; + var onCreated = function(offer) { + ccall("_emrtc_session_description_created", + "void", + ["number", "string", "string"], + [c_ptr, offer.type, offer.sdp] + ); + }; + var onSet = function() { + if (type != "offer") { + return; + } + conn.createAnswer().then(onCreated); + }; + conn.setRemoteDescription({ + "sdp": sdp, + "type": type + }).then(onSet).catch(onError); + }, _js_id, type.utf8().get_data(), sdp.utf8().get_data()); + /* clang-format on */ + return OK; +} + +Error WebRTCPeerConnectionJS::add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName) { + /* clang-format off */ + EM_ASM({ + var dict = Module.IDHandler.get($0); + var conn = dict["conn"]; + var c_ptr = dict["ptr"]; + var sdpMidName = UTF8ToString($1); + var sdpMlineIndexName = UTF8ToString($2); + var sdpName = UTF8ToString($3); + conn.addIceCandidate(new RTCIceCandidate({ + "candidate": sdpName, + "sdpMid": sdpMidName, + "sdpMlineIndex": sdpMlineIndexName + })); + }, _js_id, sdpMidName.utf8().get_data(), sdpMlineIndexName, sdpName.utf8().get_data()); + /* clang-format on */ + return OK; +} + +Error WebRTCPeerConnectionJS::initialize(Dictionary p_config) { + _emrtc_create_pc(_js_id, p_config); + return OK; +} + +Ref<WebRTCDataChannel> WebRTCPeerConnectionJS::create_data_channel(String p_channel, Dictionary p_channel_config) { + String config = JSON::print(p_channel_config); + /* clang-format off */ + int id = EM_ASM_INT({ + try { + var dict = Module.IDHandler.get($0); + if (!dict) return 0; + var label = UTF8ToString($1); + var config = JSON.parse(UTF8ToString($2)); + var conn = dict["conn"]; + return Module.IDHandler.add({ + "channel": conn.createDataChannel(label, config), + "ptr": null + }) + } catch (e) { + return 0; + } + }, _js_id, p_channel.utf8().get_data(), config.utf8().get_data()); + /* clang-format on */ + ERR_FAIL_COND_V(id == 0, NULL); + return memnew(WebRTCDataChannelJS(id)); +} + +Error WebRTCPeerConnectionJS::poll() { + return OK; +} + +WebRTCPeerConnection::ConnectionState WebRTCPeerConnectionJS::get_connection_state() const { + return _conn_state; +} + +WebRTCPeerConnectionJS::WebRTCPeerConnectionJS() { + _conn_state = STATE_NEW; + + /* clang-format off */ + _js_id = EM_ASM_INT({ + return Module.IDHandler.add({"conn": null, "ptr": $0}); + }, this); + /* clang-format on */ + Dictionary config; + _emrtc_create_pc(_js_id, config); +} + +WebRTCPeerConnectionJS::~WebRTCPeerConnectionJS() { + close(); + /* clang-format off */ + EM_ASM({ + Module.IDHandler.remove($0); + }, _js_id); + /* clang-format on */ +}; +#endif diff --git a/modules/webrtc/webrtc_peer_connection_js.h b/modules/webrtc/webrtc_peer_connection_js.h new file mode 100644 index 0000000000..43c0e3d6ee --- /dev/null +++ b/modules/webrtc/webrtc_peer_connection_js.h @@ -0,0 +1,66 @@ +/*************************************************************************/ +/* webrtc_peer_connection_js.h */ +/*************************************************************************/ +/* This file is part of: */ +/* GODOT ENGINE */ +/* https://godotengine.org */ +/*************************************************************************/ +/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur. */ +/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md) */ +/* */ +/* 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. */ +/*************************************************************************/ + +#ifndef WEBRTC_PEER_CONNECTION_JS_H +#define WEBRTC_PEER_CONNECTION_JS_H + +#ifdef JAVASCRIPT_ENABLED + +#include "webrtc_peer_connection.h" + +class WebRTCPeerConnectionJS : public WebRTCPeerConnection { + +private: + int _js_id; + ConnectionState _conn_state; + +public: + static WebRTCPeerConnection *_create() { return memnew(WebRTCPeerConnectionJS); } + static void make_default() { WebRTCPeerConnection::_create = WebRTCPeerConnectionJS::_create; } + + void _on_connection_state_changed(); + virtual ConnectionState get_connection_state() const; + + virtual Error initialize(Dictionary configuration = Dictionary()); + virtual Ref<WebRTCDataChannel> create_data_channel(String p_channel_name, Dictionary p_channel_config = Dictionary()); + virtual Error create_offer(); + virtual Error set_remote_description(String type, String sdp); + virtual Error set_local_description(String type, String sdp); + virtual Error add_ice_candidate(String sdpMidName, int sdpMlineIndexName, String sdpName); + virtual Error poll(); + virtual void close(); + + WebRTCPeerConnectionJS(); + ~WebRTCPeerConnectionJS(); +}; + +#endif + +#endif // WEBRTC_PEER_CONNECTION_JS_H |