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-rw-r--r--modules/webrtc/config.py3
-rw-r--r--modules/webrtc/doc_classes/WebRTCDataChannel.xml21
-rw-r--r--modules/webrtc/doc_classes/WebRTCMultiplayer.xml85
-rw-r--r--modules/webrtc/doc_classes/WebRTCPeerConnection.xml63
-rw-r--r--modules/webrtc/register_types.cpp2
-rw-r--r--modules/webrtc/webrtc_data_channel_js.cpp33
-rw-r--r--modules/webrtc/webrtc_multiplayer.cpp384
-rw-r--r--modules/webrtc/webrtc_multiplayer.h116
8 files changed, 697 insertions, 10 deletions
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