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-rw-r--r--modules/webrtc/doc_classes/WebRTCMultiplayerPeer.xml39
-rw-r--r--modules/webrtc/doc_classes/WebRTCPeerConnection.xml39
-rw-r--r--modules/webrtc/doc_classes/WebRTCPeerConnectionExtension.xml10
-rw-r--r--modules/webrtc/library_godot_webrtc.js196
-rw-r--r--modules/webrtc/webrtc_multiplayer_peer.cpp93
-rw-r--r--modules/webrtc/webrtc_multiplayer_peer.h43
-rw-r--r--modules/webrtc/webrtc_peer_connection.cpp13
-rw-r--r--modules/webrtc/webrtc_peer_connection.h19
-rw-r--r--modules/webrtc/webrtc_peer_connection_extension.cpp2
-rw-r--r--modules/webrtc/webrtc_peer_connection_extension.h2
-rw-r--r--modules/webrtc/webrtc_peer_connection_js.cpp23
-rw-r--r--modules/webrtc/webrtc_peer_connection_js.h14
12 files changed, 357 insertions, 136 deletions
diff --git a/modules/webrtc/doc_classes/WebRTCMultiplayerPeer.xml b/modules/webrtc/doc_classes/WebRTCMultiplayerPeer.xml
index 927888fe21..5266a36637 100644
--- a/modules/webrtc/doc_classes/WebRTCMultiplayerPeer.xml
+++ b/modules/webrtc/doc_classes/WebRTCMultiplayerPeer.xml
@@ -6,7 +6,7 @@
<description>
This class constructs a full mesh of [WebRTCPeerConnection] (one connection for each peer) that can be used as a [member MultiplayerAPI.multiplayer_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 MultiplayerPeer.connection_succeeded] and [signal MultiplayerPeer.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 [MultiplayerPeer].
+ When creating the peer via [method create_client] or [method create_server] the [method MultiplayerPeer.is_server_relay_supported] method will return [code]true[/code] enabling peer exchange and packet relaying when supported by the [MultiplayerAPI] implementation.
[b]Note:[/b] When exporting to Android, make sure to enable the [code]INTERNET[/code] permission in the Android export preset before exporting the project or using one-click deploy. Otherwise, network communication of any kind will be blocked by Android.
</description>
<tutorials>
@@ -22,10 +22,29 @@
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" />
+ <method name="create_client">
+ <return type="int" enum="Error" />
+ <param index="0" name="peer_id" type="int" />
+ <param index="1" name="channels_config" type="Array" default="[]" />
+ <description>
+ Initialize the multiplayer peer as a client with the given [code]peer_id[/code] (must be between 2 and 2147483647). In this mode, you should only call [method add_peer] once and with [code]peer_id[/code] of [code]1[/code]. This mode enables [method MultiplayerPeer.is_server_relay_supported], allowing the upper [MultiplayerAPI] layer to perform peer exchange and packet relaying.
+ You can optionally specify a [code]channels_config[/code] array of [enum MultiplayerPeer.TransferMode] which will be used to create extra channels (WebRTC only supports one transfer mode per channel).
+ </description>
+ </method>
+ <method name="create_mesh">
+ <return type="int" enum="Error" />
+ <param index="0" name="peer_id" type="int" />
+ <param index="1" name="channels_config" type="Array" default="[]" />
+ <description>
+ Initialize the multiplayer peer as a mesh (i.e. all peers connect to each other) with the given [code]peer_id[/code] (must be between 1 and 2147483647).
+ </description>
+ </method>
+ <method name="create_server">
+ <return type="int" enum="Error" />
+ <param index="0" name="channels_config" type="Array" default="[]" />
<description>
- Close all the add peer connections and channels, freeing all resources.
+ Initialize the multiplayer peer as a server (with unique ID of [code]1[/code]). This mode enables [method MultiplayerPeer.is_server_relay_supported], allowing the upper [MultiplayerAPI] layer to perform peer exchange and packet relaying.
+ You can optionally specify a [code]channels_config[/code] array of [enum MultiplayerPeer.TransferMode] which will be used to create extra channels (WebRTC only supports one transfer mode per channel).
</description>
</method>
<method name="get_peer">
@@ -48,18 +67,6 @@
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" />
- <param index="0" name="peer_id" type="int" />
- <param index="1" name="server_compatibility" type="bool" default="false" />
- <param index="2" name="channels_config" type="Array" default="[]" />
- <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 MultiplayerPeer.CONNECTION_CONNECTED] and [signal MultiplayerPeer.connection_succeeded] will not be emitted.
- If [code]server_compatibilty[/code] is [code]true[/code] the peer will suppress all [signal MultiplayerPeer.peer_connected] signals until a peer with id [constant MultiplayerPeer.TARGET_PEER_SERVER] connects and then emit [signal MultiplayerPeer.connection_succeeded]. After that the signal [signal MultiplayerPeer.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 MultiplayerPeer.server_disconnected] will be emitted and state will become [constant MultiplayerPeer.CONNECTION_CONNECTED].
- You can optionally specify a [code]channels_config[/code] array of [enum MultiplayerPeer.TransferMode] which will be used to create extra channels (WebRTC only supports one transfer mode per channel).
- </description>
- </method>
<method name="remove_peer">
<return type="void" />
<param index="0" name="peer_id" type="int" />
diff --git a/modules/webrtc/doc_classes/WebRTCPeerConnection.xml b/modules/webrtc/doc_classes/WebRTCPeerConnection.xml
index e99aeb4f51..4ecc71ddbb 100644
--- a/modules/webrtc/doc_classes/WebRTCPeerConnection.xml
+++ b/modules/webrtc/doc_classes/WebRTCPeerConnection.xml
@@ -67,6 +67,18 @@
Returns the connection state. See [enum ConnectionState].
</description>
</method>
+ <method name="get_gathering_state" qualifiers="const">
+ <return type="int" enum="WebRTCPeerConnection.GatheringState" />
+ <description>
+ Returns the ICE [enum GatheringState] of the connection. This lets you detect, for example, when collection of ICE candidates has finished.
+ </description>
+ </method>
+ <method name="get_signaling_state" qualifiers="const">
+ <return type="int" enum="WebRTCPeerConnection.SignalingState" />
+ <description>
+ Returns the [enum SignalingState] on the local end of the connection while connecting or reconnecting to another peer.
+ </description>
+ </method>
<method name="initialize">
<return type="int" enum="Error" />
<param index="0" name="configuration" type="Dictionary" default="{}" />
@@ -165,5 +177,32 @@
<constant name="STATE_CLOSED" value="5" enum="ConnectionState">
The peer connection is closed (after calling [method close] for example).
</constant>
+ <constant name="GATHERING_STATE_NEW" value="0" enum="GatheringState">
+ The peer connection was just created and hasn't done any networking yet.
+ </constant>
+ <constant name="GATHERING_STATE_GATHERING" value="1" enum="GatheringState">
+ The ICE agent is in the process of gathering candidates for the connection.
+ </constant>
+ <constant name="GATHERING_STATE_COMPLETE" value="2" enum="GatheringState">
+ The ICE agent has finished gathering candidates. If something happens that requires collecting new candidates, such as a new interface being added or the addition of a new ICE server, the state will revert to gathering to gather those candidates.
+ </constant>
+ <constant name="SIGNALING_STATE_STABLE" value="0" enum="SignalingState">
+ There is no ongoing exchange of offer and answer underway. This may mean that the [WebRTCPeerConnection] is new ([constant STATE_NEW]) or that negotiation is complete and a connection has been established ([constant STATE_CONNECTED]).
+ </constant>
+ <constant name="SIGNALING_STATE_HAVE_LOCAL_OFFER" value="1" enum="SignalingState">
+ The local peer has called [method set_local_description], passing in SDP representing an offer (usually created by calling [method create_offer]), and the offer has been applied successfully.
+ </constant>
+ <constant name="SIGNALING_STATE_HAVE_REMOTE_OFFER" value="2" enum="SignalingState">
+ The remote peer has created an offer and used the signaling server to deliver it to the local peer, which has set the offer as the remote description by calling [method set_remote_description].
+ </constant>
+ <constant name="SIGNALING_STATE_HAVE_LOCAL_PRANSWER" value="3" enum="SignalingState">
+ The offer sent by the remote peer has been applied and an answer has been created and applied by calling [method set_local_description]. This provisional answer describes the supported media formats and so forth, but may not have a complete set of ICE candidates included. Further candidates will be delivered separately later.
+ </constant>
+ <constant name="SIGNALING_STATE_HAVE_REMOTE_PRANSWER" value="4" enum="SignalingState">
+ A provisional answer has been received and successfully applied in response to an offer previously sent and established by calling [method set_local_description].
+ </constant>
+ <constant name="SIGNALING_STATE_CLOSED" value="5" enum="SignalingState">
+ The [WebRTCPeerConnection] has been closed.
+ </constant>
</constants>
</class>
diff --git a/modules/webrtc/doc_classes/WebRTCPeerConnectionExtension.xml b/modules/webrtc/doc_classes/WebRTCPeerConnectionExtension.xml
index 3c4bf18a76..474d2f6a89 100644
--- a/modules/webrtc/doc_classes/WebRTCPeerConnectionExtension.xml
+++ b/modules/webrtc/doc_classes/WebRTCPeerConnectionExtension.xml
@@ -37,6 +37,16 @@
<description>
</description>
</method>
+ <method name="_get_gathering_state" qualifiers="virtual const">
+ <return type="int" enum="WebRTCPeerConnection.GatheringState" />
+ <description>
+ </description>
+ </method>
+ <method name="_get_signaling_state" qualifiers="virtual const">
+ <return type="int" enum="WebRTCPeerConnection.SignalingState" />
+ <description>
+ </description>
+ </method>
<method name="_initialize" qualifiers="virtual">
<return type="int" enum="Error" />
<param index="0" name="p_config" type="Dictionary" />
diff --git a/modules/webrtc/library_godot_webrtc.js b/modules/webrtc/library_godot_webrtc.js
index e57e4299e0..e6604eecd7 100644
--- a/modules/webrtc/library_godot_webrtc.js
+++ b/modules/webrtc/library_godot_webrtc.js
@@ -220,64 +220,123 @@ mergeInto(LibraryManager.library, GodotRTCDataChannel);
const GodotRTCPeerConnection = {
$GodotRTCPeerConnection__deps: ['$IDHandler', '$GodotRuntime', '$GodotRTCDataChannel'],
$GodotRTCPeerConnection: {
- onstatechange: function (p_id, p_conn, callback, event) {
- const ref = IDHandler.get(p_id);
- if (!ref) {
- return;
- }
- let state;
- switch (p_conn.iceConnectionState) {
- case 'new':
- state = 0;
- break;
- case 'checking':
- state = 1;
- break;
- case 'connected':
- case 'completed':
- state = 2;
- break;
- case 'disconnected':
- state = 3;
- break;
- case 'failed':
- state = 4;
- break;
- case 'closed':
- default:
- state = 5;
- break;
- }
- callback(state);
+ // Enums
+ ConnectionState: {
+ 'new': 0,
+ 'connecting': 1,
+ 'connected': 2,
+ 'disconnected': 3,
+ 'failed': 4,
+ 'closed': 5,
},
- onicecandidate: function (p_id, callback, event) {
- const ref = IDHandler.get(p_id);
- if (!ref || !event.candidate) {
- return;
+ ConnectionStateCompat: {
+ // Using values from IceConnectionState for browsers that do not support ConnectionState (notably Firefox).
+ 'new': 0,
+ 'checking': 1,
+ 'connected': 2,
+ 'completed': 2,
+ 'disconnected': 3,
+ 'failed': 4,
+ 'closed': 5,
+ },
+
+ IceGatheringState: {
+ 'new': 0,
+ 'gathering': 1,
+ 'complete': 2,
+ },
+
+ SignalingState: {
+ 'stable': 0,
+ 'have-local-offer': 1,
+ 'have-remote-offer': 2,
+ 'have-local-pranswer': 3,
+ 'have-remote-pranswer': 4,
+ 'closed': 5,
+ },
+
+ // Callbacks
+ create: function (config, onConnectionChange, onSignalingChange, onIceGatheringChange, onIceCandidate, onDataChannel) {
+ let conn = null;
+ try {
+ conn = new RTCPeerConnection(config);
+ } catch (e) {
+ GodotRuntime.error(e);
+ return 0;
}
- const c = event.candidate;
- const candidate_str = GodotRuntime.allocString(c.candidate);
- const mid_str = GodotRuntime.allocString(c.sdpMid);
- callback(mid_str, c.sdpMLineIndex, candidate_str);
- GodotRuntime.free(candidate_str);
- GodotRuntime.free(mid_str);
+ const id = IDHandler.add(conn);
+
+ if ('connectionState' in conn && conn['connectionState'] !== undefined) {
+ // Use "connectionState" if supported
+ conn.onconnectionstatechange = function (event) {
+ if (!IDHandler.get(id)) {
+ return;
+ }
+ onConnectionChange(GodotRTCPeerConnection.ConnectionState[conn.connectionState] || 0);
+ };
+ } else {
+ // Fall back to using "iceConnectionState" when "connectionState" is not supported (notably Firefox).
+ conn.oniceconnectionstatechange = function (event) {
+ if (!IDHandler.get(id)) {
+ return;
+ }
+ onConnectionChange(GodotRTCPeerConnection.ConnectionStateCompat[conn.iceConnectionState] || 0);
+ };
+ }
+ conn.onicegatheringstatechange = function (event) {
+ if (!IDHandler.get(id)) {
+ return;
+ }
+ onIceGatheringChange(GodotRTCPeerConnection.IceGatheringState[conn.iceGatheringState] || 0);
+ };
+ conn.onsignalingstatechange = function (event) {
+ if (!IDHandler.get(id)) {
+ return;
+ }
+ onSignalingChange(GodotRTCPeerConnection.SignalingState[conn.signalingState] || 0);
+ };
+ conn.onicecandidate = function (event) {
+ if (!IDHandler.get(id)) {
+ return;
+ }
+ const c = event.candidate;
+ if (!c || !c.candidate) {
+ return;
+ }
+ const candidate_str = GodotRuntime.allocString(c.candidate);
+ const mid_str = GodotRuntime.allocString(c.sdpMid);
+ onIceCandidate(mid_str, c.sdpMLineIndex, candidate_str);
+ GodotRuntime.free(candidate_str);
+ GodotRuntime.free(mid_str);
+ };
+ conn.ondatachannel = function (event) {
+ if (!IDHandler.get(id)) {
+ return;
+ }
+ const cid = IDHandler.add(event.channel);
+ onDataChannel(cid);
+ };
+ return id;
},
- ondatachannel: function (p_id, callback, event) {
- const ref = IDHandler.get(p_id);
- if (!ref) {
+ destroy: function (p_id) {
+ const conn = IDHandler.get(p_id);
+ if (!conn) {
return;
}
-
- const cid = IDHandler.add(event.channel);
- callback(cid);
+ conn.onconnectionstatechange = null;
+ conn.oniceconnectionstatechange = null;
+ conn.onicegatheringstatechange = null;
+ conn.onsignalingstatechange = null;
+ conn.onicecandidate = null;
+ conn.ondatachannel = null;
+ IDHandler.remove(p_id);
},
onsession: function (p_id, callback, session) {
- const ref = IDHandler.get(p_id);
- if (!ref) {
+ if (!IDHandler.get(p_id)) {
return;
}
const type_str = GodotRuntime.allocString(session.type);
@@ -297,27 +356,19 @@ const GodotRTCPeerConnection = {
},
},
- godot_js_rtc_pc_create__sig: 'iiiiii',
- godot_js_rtc_pc_create: function (p_config, p_ref, p_on_state_change, p_on_candidate, p_on_datachannel) {
- const onstatechange = GodotRuntime.get_func(p_on_state_change).bind(null, p_ref);
- const oncandidate = GodotRuntime.get_func(p_on_candidate).bind(null, p_ref);
- const ondatachannel = GodotRuntime.get_func(p_on_datachannel).bind(null, p_ref);
-
- const config = JSON.parse(GodotRuntime.parseString(p_config));
- let conn = null;
- try {
- conn = new RTCPeerConnection(config);
- } catch (e) {
- GodotRuntime.error(e);
- return 0;
- }
-
- const base = GodotRTCPeerConnection;
- const id = IDHandler.add(conn);
- conn.oniceconnectionstatechange = base.onstatechange.bind(null, id, conn, onstatechange);
- conn.onicecandidate = base.onicecandidate.bind(null, id, oncandidate);
- conn.ondatachannel = base.ondatachannel.bind(null, id, ondatachannel);
- return id;
+ godot_js_rtc_pc_create__sig: 'iiiiiiii',
+ godot_js_rtc_pc_create: function (p_config, p_ref, p_on_connection_state_change, p_on_ice_gathering_state_change, p_on_signaling_state_change, p_on_ice_candidate, p_on_datachannel) {
+ const wrap = function (p_func) {
+ return GodotRuntime.get_func(p_func).bind(null, p_ref);
+ };
+ return GodotRTCPeerConnection.create(
+ JSON.parse(GodotRuntime.parseString(p_config)),
+ wrap(p_on_connection_state_change),
+ wrap(p_on_signaling_state_change),
+ wrap(p_on_ice_gathering_state_change),
+ wrap(p_on_ice_candidate),
+ wrap(p_on_datachannel)
+ );
},
godot_js_rtc_pc_close__sig: 'vi',
@@ -331,14 +382,7 @@ const GodotRTCPeerConnection = {
godot_js_rtc_pc_destroy__sig: 'vi',
godot_js_rtc_pc_destroy: function (p_id) {
- const ref = IDHandler.get(p_id);
- if (!ref) {
- return;
- }
- ref.oniceconnectionstatechange = null;
- ref.onicecandidate = null;
- ref.ondatachannel = null;
- IDHandler.remove(p_id);
+ GodotRTCPeerConnection.destroy(p_id);
},
godot_js_rtc_pc_offer_create__sig: 'viiii',
diff --git a/modules/webrtc/webrtc_multiplayer_peer.cpp b/modules/webrtc/webrtc_multiplayer_peer.cpp
index e03b6b2473..38c33a2dbc 100644
--- a/modules/webrtc/webrtc_multiplayer_peer.cpp
+++ b/modules/webrtc/webrtc_multiplayer_peer.cpp
@@ -34,13 +34,14 @@
#include "core/os/os.h"
void WebRTCMultiplayerPeer::_bind_methods() {
- ClassDB::bind_method(D_METHOD("initialize", "peer_id", "server_compatibility", "channels_config"), &WebRTCMultiplayerPeer::initialize, DEFVAL(false), DEFVAL(Array()));
+ ClassDB::bind_method(D_METHOD("create_server", "channels_config"), &WebRTCMultiplayerPeer::create_server, DEFVAL(Array()));
+ ClassDB::bind_method(D_METHOD("create_client", "peer_id", "channels_config"), &WebRTCMultiplayerPeer::create_client, DEFVAL(Array()));
+ ClassDB::bind_method(D_METHOD("create_mesh", "peer_id", "channels_config"), &WebRTCMultiplayerPeer::create_mesh, DEFVAL(Array()));
ClassDB::bind_method(D_METHOD("add_peer", "peer", "peer_id", "unreliable_lifetime"), &WebRTCMultiplayerPeer::add_peer, DEFVAL(1));
ClassDB::bind_method(D_METHOD("remove_peer", "peer_id"), &WebRTCMultiplayerPeer::remove_peer);
ClassDB::bind_method(D_METHOD("has_peer", "peer_id"), &WebRTCMultiplayerPeer::has_peer);
ClassDB::bind_method(D_METHOD("get_peer", "peer_id"), &WebRTCMultiplayerPeer::get_peer);
ClassDB::bind_method(D_METHOD("get_peers"), &WebRTCMultiplayerPeer::get_peers);
- ClassDB::bind_method(D_METHOD("close"), &WebRTCMultiplayerPeer::close);
}
void WebRTCMultiplayerPeer::set_target_peer(int p_peer_id) {
@@ -52,6 +53,15 @@ int WebRTCMultiplayerPeer::get_packet_peer() const {
return next_packet_peer;
}
+int WebRTCMultiplayerPeer::get_packet_channel() const {
+ return next_packet_channel < CH_RESERVED_MAX ? 0 : next_packet_channel - CH_RESERVED_MAX + 1;
+}
+
+MultiplayerPeer::TransferMode WebRTCMultiplayerPeer::get_packet_mode() const {
+ ERR_FAIL_INDEX_V(next_packet_channel, channels_modes.size(), TRANSFER_MODE_RELIABLE);
+ return channels_modes[next_packet_channel];
+}
+
bool WebRTCMultiplayerPeer::is_server() const {
return unique_id == TARGET_PEER_SERVER;
}
@@ -113,24 +123,14 @@ void WebRTCMultiplayerPeer::poll() {
// Signal newly connected peers
for (int &E : add) {
// Already connected to server: simply notify new peer.
- // NOTE: Mesh is always connected.
- if (connection_status == CONNECTION_CONNECTED) {
- emit_signal(SNAME("peer_connected"), E);
- }
-
- // Server emulation mode suppresses peer_conencted until server connects.
- if (server_compat && E == TARGET_PEER_SERVER) {
+ if (network_mode == MODE_CLIENT) {
+ ERR_CONTINUE(E != TARGET_PEER_SERVER); // Bug.
// Server connected.
connection_status = CONNECTION_CONNECTED;
emit_signal(SNAME("peer_connected"), TARGET_PEER_SERVER);
emit_signal(SNAME("connection_succeeded"));
- // Notify of all previously connected peers
- for (const KeyValue<int, Ref<ConnectedPeer>> &F : peer_map) {
- if (F.key != 1 && F.value->connected) {
- emit_signal(SNAME("peer_connected"), F.key);
- }
- }
- break; // Because we already notified of all newly added peers.
+ } else {
+ emit_signal(SNAME("peer_connected"), E);
}
}
// Fetch next packet
@@ -150,11 +150,14 @@ void WebRTCMultiplayerPeer::_find_next_peer() {
++E;
continue;
}
+ int idx = 0;
for (const Ref<WebRTCDataChannel> &F : E->value->channels) {
if (F->get_available_packet_count()) {
+ next_packet_channel = idx;
next_packet_peer = E->key;
return;
}
+ idx++;
}
++E;
}
@@ -165,11 +168,14 @@ void WebRTCMultiplayerPeer::_find_next_peer() {
++E;
continue;
}
+ int idx = 0;
for (const Ref<WebRTCDataChannel> &F : E->value->channels) {
if (F->get_available_packet_count()) {
+ next_packet_channel = idx;
next_packet_peer = E->key;
return;
}
+ idx++;
}
if (E->key == (int)next_packet_peer) {
break;
@@ -177,6 +183,7 @@ void WebRTCMultiplayerPeer::_find_next_peer() {
++E;
}
// No packet found
+ next_packet_channel = 0;
next_packet_peer = 0;
}
@@ -184,11 +191,28 @@ MultiplayerPeer::ConnectionStatus WebRTCMultiplayerPeer::get_connection_status()
return connection_status;
}
-Error WebRTCMultiplayerPeer::initialize(int p_self_id, bool p_server_compat, Array p_channels_config) {
+Error WebRTCMultiplayerPeer::create_server(Array p_channels_config) {
+ return _initialize(1, MODE_SERVER, p_channels_config);
+}
+
+Error WebRTCMultiplayerPeer::create_client(int p_self_id, Array p_channels_config) {
+ ERR_FAIL_COND_V_MSG(p_self_id == 1, ERR_INVALID_PARAMETER, "Clients cannot have ID 1.");
+ return _initialize(p_self_id, MODE_CLIENT, p_channels_config);
+}
+
+Error WebRTCMultiplayerPeer::create_mesh(int p_self_id, Array p_channels_config) {
+ return _initialize(p_self_id, MODE_MESH, p_channels_config);
+}
+
+Error WebRTCMultiplayerPeer::_initialize(int p_self_id, NetworkMode p_mode, Array p_channels_config) {
ERR_FAIL_COND_V(p_self_id < 1 || p_self_id > ~(1 << 31), ERR_INVALID_PARAMETER);
channels_config.clear();
+ channels_modes.clear();
+ channels_modes.push_back(TRANSFER_MODE_RELIABLE);
+ channels_modes.push_back(TRANSFER_MODE_UNRELIABLE_ORDERED);
+ channels_modes.push_back(TRANSFER_MODE_UNRELIABLE);
for (int i = 0; i < p_channels_config.size(); i++) {
- ERR_FAIL_COND_V_MSG(p_channels_config[i].get_type() != Variant::INT, ERR_INVALID_PARAMETER, "The 'channels_config' array must contain only enum values from 'MultiplayerPeer.Multiplayer::TransferMode'");
+ ERR_FAIL_COND_V_MSG(p_channels_config[i].get_type() != Variant::INT, ERR_INVALID_PARAMETER, "The 'channels_config' array must contain only enum values from 'MultiplayerPeer.TransferMode'");
int mode = p_channels_config[i].operator int();
// Initialize data channel configurations.
Dictionary cfg;
@@ -207,16 +231,17 @@ Error WebRTCMultiplayerPeer::initialize(int p_self_id, bool p_server_compat, Arr
case TRANSFER_MODE_RELIABLE:
break;
default:
- ERR_FAIL_V_MSG(ERR_INVALID_PARAMETER, vformat("The 'channels_config' array must contain only enum values from 'MultiplayerPeer.Multiplayer::TransferMode'. Got: %d", mode));
+ ERR_FAIL_V_MSG(ERR_INVALID_PARAMETER, vformat("The 'channels_config' array must contain only enum values from 'MultiplayerPeer.TransferMode'. Got: %d", mode));
}
channels_config.push_back(cfg);
+ channels_modes.push_back((TransferMode)mode);
}
unique_id = p_self_id;
- server_compat = p_server_compat;
+ network_mode = p_mode;
// Mesh and server are always connected
- if (!server_compat || p_self_id == 1) {
+ if (p_mode != MODE_CLIENT) {
connection_status = CONNECTION_CONNECTED;
} else {
connection_status = CONNECTION_CONNECTING;
@@ -224,6 +249,10 @@ Error WebRTCMultiplayerPeer::initialize(int p_self_id, bool p_server_compat, Arr
return OK;
}
+bool WebRTCMultiplayerPeer::is_server_relay_supported() const {
+ return network_mode == MODE_SERVER || network_mode == MODE_CLIENT;
+}
+
int WebRTCMultiplayerPeer::get_unique_id() const {
ERR_FAIL_COND_V(connection_status == CONNECTION_DISCONNECTED, 1);
return unique_id;
@@ -261,7 +290,10 @@ Dictionary WebRTCMultiplayerPeer::get_peers() {
}
Error WebRTCMultiplayerPeer::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(network_mode == MODE_NONE, ERR_UNCONFIGURED);
+ ERR_FAIL_COND_V(network_mode == MODE_CLIENT && p_peer_id != 1, ERR_INVALID_PARAMETER);
+ ERR_FAIL_COND_V(network_mode == MODE_SERVER && p_peer_id == 1, ERR_INVALID_PARAMETER);
+ ERR_FAIL_COND_V(p_peer_id < 1 || p_peer_id > ~(1 << 31), ERR_INVALID_PARAMETER);
ERR_FAIL_COND_V(p_unreliable_lifetime < 0, ERR_INVALID_PARAMETER);
ERR_FAIL_COND_V(is_refusing_new_connections(), ERR_UNAUTHORIZED);
// Peer must be valid, and in new state (to create data channels)
@@ -308,10 +340,21 @@ void WebRTCMultiplayerPeer::remove_peer(int p_peer_id) {
if (peer->connected) {
peer->connected = false;
emit_signal(SNAME("peer_disconnected"), p_peer_id);
- if (server_compat && p_peer_id == TARGET_PEER_SERVER) {
- emit_signal(SNAME("server_disconnected"));
+ if (network_mode == MODE_CLIENT && p_peer_id == TARGET_PEER_SERVER) {
+ connection_status = CONNECTION_DISCONNECTED;
+ }
+ }
+}
+
+void WebRTCMultiplayerPeer::disconnect_peer(int p_peer_id, bool p_force) {
+ ERR_FAIL_COND(!peer_map.has(p_peer_id));
+ if (p_force) {
+ peer_map.erase(p_peer_id);
+ if (network_mode == MODE_CLIENT && p_peer_id == TARGET_PEER_SERVER) {
connection_status = CONNECTION_DISCONNECTED;
}
+ } else {
+ peer_map[p_peer_id]->connection->close(); // Will be removed during next poll.
}
}
@@ -403,7 +446,9 @@ void WebRTCMultiplayerPeer::close() {
channels_config.clear();
unique_id = 0;
next_packet_peer = 0;
+ next_packet_channel = 0;
target_peer = 0;
+ network_mode = MODE_NONE;
connection_status = CONNECTION_DISCONNECTED;
}
diff --git a/modules/webrtc/webrtc_multiplayer_peer.h b/modules/webrtc/webrtc_multiplayer_peer.h
index ea7c60036b..0556ef029c 100644
--- a/modules/webrtc/webrtc_multiplayer_peer.h
+++ b/modules/webrtc/webrtc_multiplayer_peer.h
@@ -48,6 +48,13 @@ private:
CH_RESERVED_MAX = 3
};
+ enum NetworkMode {
+ MODE_NONE,
+ MODE_SERVER,
+ MODE_CLIENT,
+ MODE_MESH,
+ };
+
class ConnectedPeer : public RefCounted {
public:
Ref<WebRTCPeerConnection> connection;
@@ -67,43 +74,53 @@ private:
int client_count = 0;
ConnectionStatus connection_status = CONNECTION_DISCONNECTED;
int next_packet_peer = 0;
- bool server_compat = false;
+ int next_packet_channel = 0;
+ NetworkMode network_mode = MODE_NONE;
HashMap<int, Ref<ConnectedPeer>> peer_map;
+ List<TransferMode> channels_modes;
List<Dictionary> channels_config;
void _peer_to_dict(Ref<ConnectedPeer> p_connected_peer, Dictionary &r_dict);
void _find_next_peer();
+ Ref<ConnectedPeer> _get_next_peer();
+ Error _initialize(int p_self_id, NetworkMode p_mode, Array p_channels_config = Array());
public:
WebRTCMultiplayerPeer() {}
~WebRTCMultiplayerPeer();
- Error initialize(int p_self_id, bool p_server_compat = false, Array p_channels_config = Array());
+ Error create_server(Array p_channels_config = Array());
+ Error create_client(int p_self_id, Array p_channels_config = Array());
+ Error create_mesh(int p_self_id, Array p_channels_config = Array());
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) override; ///< buffer is GONE after next get_packet
- Error put_packet(const uint8_t *p_buffer, int p_buffer_size) override;
- int get_available_packet_count() const override;
- int get_max_packet_size() const override;
+ virtual Error get_packet(const uint8_t **r_buffer, int &r_buffer_size) override; ///< buffer is GONE after next get_packet
+ virtual Error put_packet(const uint8_t *p_buffer, int p_buffer_size) override;
+ virtual int get_available_packet_count() const override;
+ virtual int get_max_packet_size() const override;
// MultiplayerPeer
- void set_target_peer(int p_peer_id) override;
+ virtual void set_target_peer(int p_peer_id) override;
- int get_unique_id() const override;
- int get_packet_peer() const override;
+ virtual int get_unique_id() const override;
+ virtual int get_packet_peer() const override;
+ virtual int get_packet_channel() const override;
+ virtual TransferMode get_packet_mode() const override;
- bool is_server() const override;
+ virtual bool is_server() const override;
+ virtual bool is_server_relay_supported() const override;
- void poll() override;
+ virtual void poll() override;
+ virtual void close() override;
+ virtual void disconnect_peer(int p_peer_id, bool p_force = false) override;
- ConnectionStatus get_connection_status() const override;
+ virtual ConnectionStatus get_connection_status() const override;
};
#endif // WEBRTC_MULTIPLAYER_PEER_H
diff --git a/modules/webrtc/webrtc_peer_connection.cpp b/modules/webrtc/webrtc_peer_connection.cpp
index d885b9262b..5aa891d35c 100644
--- a/modules/webrtc/webrtc_peer_connection.cpp
+++ b/modules/webrtc/webrtc_peer_connection.cpp
@@ -69,6 +69,8 @@ void WebRTCPeerConnection::_bind_methods() {
ClassDB::bind_method(D_METHOD("close"), &WebRTCPeerConnection::close);
ClassDB::bind_method(D_METHOD("get_connection_state"), &WebRTCPeerConnection::get_connection_state);
+ ClassDB::bind_method(D_METHOD("get_gathering_state"), &WebRTCPeerConnection::get_gathering_state);
+ ClassDB::bind_method(D_METHOD("get_signaling_state"), &WebRTCPeerConnection::get_signaling_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")));
@@ -80,6 +82,17 @@ void WebRTCPeerConnection::_bind_methods() {
BIND_ENUM_CONSTANT(STATE_DISCONNECTED);
BIND_ENUM_CONSTANT(STATE_FAILED);
BIND_ENUM_CONSTANT(STATE_CLOSED);
+
+ BIND_ENUM_CONSTANT(GATHERING_STATE_NEW);
+ BIND_ENUM_CONSTANT(GATHERING_STATE_GATHERING);
+ BIND_ENUM_CONSTANT(GATHERING_STATE_COMPLETE);
+
+ BIND_ENUM_CONSTANT(SIGNALING_STATE_STABLE);
+ BIND_ENUM_CONSTANT(SIGNALING_STATE_HAVE_LOCAL_OFFER);
+ BIND_ENUM_CONSTANT(SIGNALING_STATE_HAVE_REMOTE_OFFER);
+ BIND_ENUM_CONSTANT(SIGNALING_STATE_HAVE_LOCAL_PRANSWER);
+ BIND_ENUM_CONSTANT(SIGNALING_STATE_HAVE_REMOTE_PRANSWER);
+ BIND_ENUM_CONSTANT(SIGNALING_STATE_CLOSED);
}
WebRTCPeerConnection::WebRTCPeerConnection() {
diff --git a/modules/webrtc/webrtc_peer_connection.h b/modules/webrtc/webrtc_peer_connection.h
index 122ea3d00f..76f29f9d68 100644
--- a/modules/webrtc/webrtc_peer_connection.h
+++ b/modules/webrtc/webrtc_peer_connection.h
@@ -47,6 +47,21 @@ public:
STATE_CLOSED
};
+ enum GatheringState {
+ GATHERING_STATE_NEW,
+ GATHERING_STATE_GATHERING,
+ GATHERING_STATE_COMPLETE,
+ };
+
+ enum SignalingState {
+ SIGNALING_STATE_STABLE,
+ SIGNALING_STATE_HAVE_LOCAL_OFFER,
+ SIGNALING_STATE_HAVE_REMOTE_OFFER,
+ SIGNALING_STATE_HAVE_LOCAL_PRANSWER,
+ SIGNALING_STATE_HAVE_REMOTE_PRANSWER,
+ SIGNALING_STATE_CLOSED,
+ };
+
private:
static StringName default_extension;
@@ -57,6 +72,8 @@ public:
static void set_default_extension(const StringName &p_name);
virtual ConnectionState get_connection_state() const = 0;
+ virtual GatheringState get_gathering_state() const = 0;
+ virtual SignalingState get_signaling_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;
@@ -74,5 +91,7 @@ public:
};
VARIANT_ENUM_CAST(WebRTCPeerConnection::ConnectionState);
+VARIANT_ENUM_CAST(WebRTCPeerConnection::GatheringState);
+VARIANT_ENUM_CAST(WebRTCPeerConnection::SignalingState);
#endif // WEBRTC_PEER_CONNECTION_H
diff --git a/modules/webrtc/webrtc_peer_connection_extension.cpp b/modules/webrtc/webrtc_peer_connection_extension.cpp
index 54143e4b79..592a1f8a97 100644
--- a/modules/webrtc/webrtc_peer_connection_extension.cpp
+++ b/modules/webrtc/webrtc_peer_connection_extension.cpp
@@ -32,6 +32,8 @@
void WebRTCPeerConnectionExtension::_bind_methods() {
GDVIRTUAL_BIND(_get_connection_state);
+ GDVIRTUAL_BIND(_get_gathering_state);
+ GDVIRTUAL_BIND(_get_signaling_state);
GDVIRTUAL_BIND(_initialize, "p_config");
GDVIRTUAL_BIND(_create_data_channel, "p_label", "p_config");
GDVIRTUAL_BIND(_create_offer);
diff --git a/modules/webrtc/webrtc_peer_connection_extension.h b/modules/webrtc/webrtc_peer_connection_extension.h
index 0c324ca45f..085069debb 100644
--- a/modules/webrtc/webrtc_peer_connection_extension.h
+++ b/modules/webrtc/webrtc_peer_connection_extension.h
@@ -53,6 +53,8 @@ public:
/** GDExtension **/
EXBIND0RC(ConnectionState, get_connection_state);
+ EXBIND0RC(GatheringState, get_gathering_state);
+ EXBIND0RC(SignalingState, get_signaling_state);
EXBIND1R(Error, initialize, Dictionary);
EXBIND0R(Error, create_offer);
EXBIND2R(Error, set_remote_description, String, String);
diff --git a/modules/webrtc/webrtc_peer_connection_js.cpp b/modules/webrtc/webrtc_peer_connection_js.cpp
index f48705253b..a371312ae9 100644
--- a/modules/webrtc/webrtc_peer_connection_js.cpp
+++ b/modules/webrtc/webrtc_peer_connection_js.cpp
@@ -51,6 +51,16 @@ void WebRTCPeerConnectionJS::_on_connection_state_changed(void *p_obj, int p_sta
peer->_conn_state = (ConnectionState)p_state;
}
+void WebRTCPeerConnectionJS::_on_gathering_state_changed(void *p_obj, int p_state) {
+ WebRTCPeerConnectionJS *peer = static_cast<WebRTCPeerConnectionJS *>(p_obj);
+ peer->_gathering_state = (GatheringState)p_state;
+}
+
+void WebRTCPeerConnectionJS::_on_signaling_state_changed(void *p_obj, int p_state) {
+ WebRTCPeerConnectionJS *peer = static_cast<WebRTCPeerConnectionJS *>(p_obj);
+ peer->_signaling_state = (SignalingState)p_state;
+}
+
void WebRTCPeerConnectionJS::_on_error(void *p_obj) {
ERR_PRINT("RTCPeerConnection error!");
}
@@ -100,7 +110,7 @@ Error WebRTCPeerConnectionJS::initialize(Dictionary p_config) {
_conn_state = STATE_NEW;
String config = Variant(p_config).to_json_string();
- _js_id = godot_js_rtc_pc_create(config.utf8().get_data(), this, &_on_connection_state_changed, &_on_ice_candidate, &_on_data_channel);
+ _js_id = godot_js_rtc_pc_create(config.utf8().get_data(), this, &_on_connection_state_changed, &_on_gathering_state_changed, &_on_signaling_state_changed, &_on_ice_candidate, &_on_data_channel);
return _js_id ? OK : FAILED;
}
@@ -117,14 +127,19 @@ Error WebRTCPeerConnectionJS::poll() {
return OK;
}
+WebRTCPeerConnection::GatheringState WebRTCPeerConnectionJS::get_gathering_state() const {
+ return _gathering_state;
+}
+
+WebRTCPeerConnection::SignalingState WebRTCPeerConnectionJS::get_signaling_state() const {
+ return _signaling_state;
+}
+
WebRTCPeerConnection::ConnectionState WebRTCPeerConnectionJS::get_connection_state() const {
return _conn_state;
}
WebRTCPeerConnectionJS::WebRTCPeerConnectionJS() {
- _conn_state = STATE_NEW;
- _js_id = 0;
-
Dictionary config;
initialize(config);
}
diff --git a/modules/webrtc/webrtc_peer_connection_js.h b/modules/webrtc/webrtc_peer_connection_js.h
index 50266129e4..e62ad6af28 100644
--- a/modules/webrtc/webrtc_peer_connection_js.h
+++ b/modules/webrtc/webrtc_peer_connection_js.h
@@ -37,11 +37,13 @@
extern "C" {
typedef void (*RTCOnIceConnectionStateChange)(void *p_obj, int p_state);
+typedef void (*RTCOnIceGatheringStateChange)(void *p_obj, int p_state);
+typedef void (*RTCOnSignalingStateChange)(void *p_obj, int p_state);
typedef void (*RTCOnIceCandidate)(void *p_obj, const char *p_mid, int p_mline_idx, const char *p_candidate);
typedef void (*RTCOnDataChannel)(void *p_obj, int p_id);
typedef void (*RTCOnSession)(void *p_obj, const char *p_type, const char *p_sdp);
typedef void (*RTCOnError)(void *p_obj);
-extern int godot_js_rtc_pc_create(const char *p_config, void *p_obj, RTCOnIceConnectionStateChange p_on_state_change, RTCOnIceCandidate p_on_candidate, RTCOnDataChannel p_on_datachannel);
+extern int godot_js_rtc_pc_create(const char *p_config, void *p_obj, RTCOnIceConnectionStateChange p_on_connection_state_change, RTCOnIceGatheringStateChange p_on_gathering_state_change, RTCOnSignalingStateChange p_on_signaling_state_change, RTCOnIceCandidate p_on_candidate, RTCOnDataChannel p_on_datachannel);
extern void godot_js_rtc_pc_close(int p_id);
extern void godot_js_rtc_pc_destroy(int p_id);
extern void godot_js_rtc_pc_offer_create(int p_id, void *p_obj, RTCOnSession p_on_session, RTCOnError p_on_error);
@@ -55,10 +57,14 @@ class WebRTCPeerConnectionJS : public WebRTCPeerConnection {
GDCLASS(WebRTCPeerConnectionJS, WebRTCPeerConnection);
private:
- int _js_id;
- ConnectionState _conn_state;
+ int _js_id = 0;
+ ConnectionState _conn_state = STATE_NEW;
+ GatheringState _gathering_state = GATHERING_STATE_NEW;
+ SignalingState _signaling_state = SIGNALING_STATE_STABLE;
static void _on_connection_state_changed(void *p_obj, int p_state);
+ static void _on_gathering_state_changed(void *p_obj, int p_state);
+ static void _on_signaling_state_changed(void *p_obj, int p_state);
static void _on_ice_candidate(void *p_obj, const char *p_mid_name, int p_mline_idx, const char *p_candidate);
static void _on_data_channel(void *p_obj, int p_channel);
static void _on_session_created(void *p_obj, const char *p_type, const char *p_session);
@@ -66,6 +72,8 @@ private:
public:
virtual ConnectionState get_connection_state() const override;
+ virtual GatheringState get_gathering_state() const override;
+ virtual SignalingState get_signaling_state() const override;
virtual Error initialize(Dictionary configuration = Dictionary()) override;
virtual Ref<WebRTCDataChannel> create_data_channel(String p_channel_name, Dictionary p_channel_config = Dictionary()) override;
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-28 06:36:58 +0000