diff options
Diffstat (limited to 'modules')
| -rw-r--r-- | modules/csg/csg_shape.h | 2 | ||||
| -rw-r--r-- | modules/recast/navigation_mesh_generator.cpp | 18 | ||||
| -rw-r--r-- | modules/webrtc/config.py | 3 | ||||
| -rw-r--r-- | modules/webrtc/doc_classes/WebRTCDataChannel.xml | 21 | ||||
| -rw-r--r-- | modules/webrtc/doc_classes/WebRTCMultiplayer.xml | 85 | ||||
| -rw-r--r-- | modules/webrtc/doc_classes/WebRTCPeerConnection.xml | 63 | ||||
| -rw-r--r-- | modules/webrtc/register_types.cpp | 2 | ||||
| -rw-r--r-- | modules/webrtc/webrtc_data_channel_js.cpp | 33 | ||||
| -rw-r--r-- | modules/webrtc/webrtc_multiplayer.cpp | 384 | ||||
| -rw-r--r-- | modules/webrtc/webrtc_multiplayer.h | 116 |
10 files changed, 716 insertions, 11 deletions
diff --git a/modules/csg/csg_shape.h b/modules/csg/csg_shape.h index a5b2238e6b..2171f27f96 100644 --- a/modules/csg/csg_shape.h +++ b/modules/csg/csg_shape.h @@ -116,9 +116,9 @@ protected: virtual void _validate_property(PropertyInfo &property) const; +public: Array get_meshes() const; -public: void set_operation(Operation p_operation); Operation get_operation() const; diff --git a/modules/recast/navigation_mesh_generator.cpp b/modules/recast/navigation_mesh_generator.cpp index 0cac07e3e7..14467dc5c7 100644 --- a/modules/recast/navigation_mesh_generator.cpp +++ b/modules/recast/navigation_mesh_generator.cpp @@ -45,6 +45,10 @@ #include "scene/resources/shape.h" #include "scene/resources/sphere_shape.h" +#ifdef MODULE_CSG_ENABLED +#include "modules/csg/csg_shape.h" +#endif + EditorNavigationMeshGenerator *EditorNavigationMeshGenerator::singleton = NULL; void EditorNavigationMeshGenerator::_add_vertex(const Vector3 &p_vec3, Vector<float> &p_verticies) { @@ -134,6 +138,20 @@ void EditorNavigationMeshGenerator::_parse_geometry(Transform p_accumulated_tran } } +#ifdef MODULE_CSG_ENABLED + if (Object::cast_to<CSGShape>(p_node) && p_generate_from != NavigationMesh::PARSED_GEOMETRY_STATIC_COLLIDERS) { + + CSGShape *csg_shape = Object::cast_to<CSGShape>(p_node); + Array meshes = csg_shape->get_meshes(); + if (!meshes.empty()) { + Ref<Mesh> mesh = meshes[1]; + if (mesh.is_valid()) { + _add_mesh(mesh, p_accumulated_transform * csg_shape->get_transform(), p_verticies, p_indices); + } + } + } +#endif + if (Object::cast_to<StaticBody>(p_node) && p_generate_from != NavigationMesh::PARSED_GEOMETRY_MESH_INSTANCES) { StaticBody *static_body = Object::cast_to<StaticBody>(p_node); diff --git a/modules/webrtc/config.py b/modules/webrtc/config.py index 2e3a18ad0e..48b4c33c5d 100644 --- a/modules/webrtc/config.py +++ b/modules/webrtc/config.py @@ -7,7 +7,8 @@ def configure(env): def get_doc_classes(): return [ "WebRTCPeerConnection", - "WebRTCDataChannel" + "WebRTCDataChannel", + "WebRTCMultiplayer" ] def get_doc_path(): diff --git a/modules/webrtc/doc_classes/WebRTCDataChannel.xml b/modules/webrtc/doc_classes/WebRTCDataChannel.xml index dcc14d4ddb..f03ae864f8 100644 --- a/modules/webrtc/doc_classes/WebRTCDataChannel.xml +++ b/modules/webrtc/doc_classes/WebRTCDataChannel.xml @@ -11,85 +11,106 @@ <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 maxPacketLifeTime 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 maxRetransmits 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-proctocol 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 WebRTCDataChannel.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 transfered as text. See [property write_mode]. </description> </method> </methods> <members> <member name="write_mode" type="int" setter="set_write_mode" getter="get_write_mode" enum="WebRTCDataChannel.WriteMode"> + 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 arraybuffer 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..2b0622fffa --- /dev/null +++ b/modules/webrtc/doc_classes/WebRTCMultiplayer.xml @@ -0,0 +1,85 @@ +<?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> + <constants> + </constants> +</class> diff --git a/modules/webrtc/doc_classes/WebRTCPeerConnection.xml b/modules/webrtc/doc_classes/WebRTCPeerConnection.xml index 8b14c60deb..aa2c856b6e 100644 --- a/modules/webrtc/doc_classes/WebRTCPeerConnection.xml +++ b/modules/webrtc/doc_classes/WebRTCPeerConnection.xml @@ -1,8 +1,17 @@ <?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 trival 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]B[/code]). + - [code]A[/code] and [code]B[/code] then generates and exchange ICE candiates 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> @@ -17,12 +26,14 @@ <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"> @@ -35,18 +46,38 @@ }"> </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: + [code] + { + "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. + } + [/code] + NOTE: 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 [code]OK[/code], [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"> @@ -57,12 +88,29 @@ }"> </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: + [code] + { + "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. + } + ] + } + [/code] </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._fixed_process]) to properly receive signals. </description> </method> <method name="set_local_description"> @@ -73,6 +121,8 @@ <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"> @@ -83,6 +133,9 @@ <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> @@ -91,6 +144,8 @@ <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"> @@ -101,6 +156,7 @@ <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"> @@ -109,21 +165,28 @@ <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, non 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 index 44e072cc89..58b68d926b 100644 --- a/modules/webrtc/register_types.cpp +++ b/modules/webrtc/register_types.cpp @@ -40,6 +40,7 @@ #include "webrtc_data_channel_gdnative.h" #include "webrtc_peer_connection_gdnative.h" #endif +#include "webrtc_multiplayer.h" void register_webrtc_types() { #ifdef JAVASCRIPT_ENABLED @@ -54,6 +55,7 @@ void register_webrtc_types() { ClassDB::register_class<WebRTCDataChannelGDNative>(); #endif ClassDB::register_virtual_class<WebRTCDataChannel>(); + ClassDB::register_class<WebRTCMultiplayer>(); } void unregister_webrtc_types() {} diff --git a/modules/webrtc/webrtc_data_channel_js.cpp b/modules/webrtc/webrtc_data_channel_js.cpp index 2e7c64aa72..ce2fada634 100644 --- a/modules/webrtc/webrtc_data_channel_js.cpp +++ b/modules/webrtc/webrtc_data_channel_js.cpp @@ -205,30 +205,45 @@ String WebRTCDataChannelJS::get_label() const { } /* clang-format off */ -#define _JS_GET(PROP) \ +#define _JS_GET(PROP, DEF) \ EM_ASM_INT({ \ var dict = Module.IDHandler.get($0); \ if (!dict || !dict["channel"]) { \ - return 0; \ - }; \ - return dict["channel"].PROP; \ + 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); + return _JS_GET(ordered, true); } int WebRTCDataChannelJS::get_id() const { - return _JS_GET(id); + return _JS_GET(id, 65535); } int WebRTCDataChannelJS::get_max_packet_life_time() const { - return _JS_GET(maxPacketLifeTime); + // 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); + return _JS_GET(maxRetransmits, 65535); } String WebRTCDataChannelJS::get_protocol() const { @@ -236,7 +251,7 @@ String WebRTCDataChannelJS::get_protocol() const { } bool WebRTCDataChannelJS::is_negotiated() const { - return _JS_GET(negotiated); + return _JS_GET(negotiated, false); } WebRTCDataChannelJS::WebRTCDataChannelJS() { diff --git a/modules/webrtc/webrtc_multiplayer.cpp b/modules/webrtc/webrtc_multiplayer.cpp new file mode 100644 index 0000000000..17dafff93a --- /dev/null +++ b/modules/webrtc/webrtc_multiplayer.cpp @@ -0,0 +1,384 @@ +/*************************************************************************/ +/* 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); + if (!E) { + ERR_EXPLAIN("Invalid Target Peer: " + itos(target_peer)); + ERR_FAIL_V(ERR_INVALID_PARAMETER); + } + 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 |