diff options
Diffstat (limited to 'modules/webrtc/doc_classes')
5 files changed, 87 insertions, 31 deletions
diff --git a/modules/webrtc/doc_classes/WebRTCDataChannel.xml b/modules/webrtc/doc_classes/WebRTCDataChannel.xml index a9ba8a23de..a186631ca8 100644 --- a/modules/webrtc/doc_classes/WebRTCDataChannel.xml +++ b/modules/webrtc/doc_classes/WebRTCDataChannel.xml @@ -22,8 +22,8 @@ <method name="get_id" qualifiers="const"> <return type="int" /> <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). + 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"> diff --git a/modules/webrtc/doc_classes/WebRTCDataChannelExtension.xml b/modules/webrtc/doc_classes/WebRTCDataChannelExtension.xml index 5387deaa47..a10ea25b8c 100644 --- a/modules/webrtc/doc_classes/WebRTCDataChannelExtension.xml +++ b/modules/webrtc/doc_classes/WebRTCDataChannelExtension.xml @@ -48,7 +48,7 @@ </description> </method> <method name="_get_packet" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <param index="0" name="r_buffer" type="const uint8_t **" /> <param index="1" name="r_buffer_size" type="int32_t*" /> <description> @@ -60,12 +60,12 @@ </description> </method> <method name="_get_ready_state" qualifiers="virtual const"> - <return type="int" /> + <return type="int" enum="WebRTCDataChannel.ChannelState" /> <description> </description> </method> <method name="_get_write_mode" qualifiers="virtual const"> - <return type="int" /> + <return type="int" enum="WebRTCDataChannel.WriteMode" /> <description> </description> </method> @@ -80,12 +80,12 @@ </description> </method> <method name="_poll" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <description> </description> </method> <method name="_put_packet" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <param index="0" name="p_buffer" type="const uint8_t*" /> <param index="1" name="p_buffer_size" type="int" /> <description> @@ -93,7 +93,7 @@ </method> <method name="_set_write_mode" qualifiers="virtual"> <return type="void" /> - <param index="0" name="p_write_mode" type="int" /> + <param index="0" name="p_write_mode" type="int" enum="WebRTCDataChannel.WriteMode" /> <description> </description> </method> 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 e22e939a66..474d2f6a89 100644 --- a/modules/webrtc/doc_classes/WebRTCPeerConnectionExtension.xml +++ b/modules/webrtc/doc_classes/WebRTCPeerConnectionExtension.xml @@ -8,7 +8,7 @@ </tutorials> <methods> <method name="_add_ice_candidate" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <param index="0" name="p_sdp_mid_name" type="String" /> <param index="1" name="p_sdp_mline_index" type="int" /> <param index="2" name="p_sdp_name" type="String" /> @@ -28,35 +28,45 @@ </description> </method> <method name="_create_offer" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <description> </description> </method> <method name="_get_connection_state" qualifiers="virtual const"> - <return type="int" /> + <return type="int" enum="WebRTCPeerConnection.ConnectionState" /> + <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" /> + <return type="int" enum="Error" /> <param index="0" name="p_config" type="Dictionary" /> <description> </description> </method> <method name="_poll" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <description> </description> </method> <method name="_set_local_description" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <param index="0" name="p_type" type="String" /> <param index="1" name="p_sdp" type="String" /> <description> </description> </method> <method name="_set_remote_description" qualifiers="virtual"> - <return type="int" /> + <return type="int" enum="Error" /> <param index="0" name="p_type" type="String" /> <param index="1" name="p_sdp" type="String" /> <description> |