WebRTC refactor. Data channels, STUN/TURN support.

A big refactor to the WebRTC module. API is now considered quite stable.

Highlights:

- Renamed `WebRTCPeer` to `WebRTCPeerConnection`.
- `WebRTCPeerConnection` no longer act as `PacketPeer`, it only handle the connection itself (a bit like `TCP_Server`)
- Added new `WebRTCDataChannel` class which inherits from `PacketPeer` to handle data transfer.
- Add `WebRTCPeerConnection.initialize` method to create a new connection with the desired configuration provided as dictionary ([see MDN docs](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/RTCPeerConnection#RTCConfiguration_dictionary)).
- Add `WebRTCPeerConnection.create_data_channel` method to create a data channel for the given connection. The connection must be in `STATE_NEW` as specified by the standard ([see MDN docs for options](https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection/createDataChannel#RTCDataChannelInit_dictionary)).
- Add a `data_channel_received` signal to `WebRTCPeerConnection` for in-band (not negotiated) channels.
- Renamed `WebRTCPeerConnection` `offer_created` signal to `session_description_created`.
- Renamed `WebRTCPeerConnection` `new_ice_candidate` signal to `ice_candidate_created`

1 files changed, 85 insertions, 0 deletions

diff --git a/modules/webrtc/webrtc_data_channel.h b/modules/webrtc/webrtc_data_channel.h
new file mode 100644
index 0000000000..0b161da784
--- /dev/null
+++ b/modules/webrtc/webrtc_data_channel.h
@@ -0,0 +1,85 @@
+/*************************************************************************/
+/*  webrtc_data_channel.h                                                */
+/*************************************************************************/
+/*                       This file is part of:                           */
+/*                           GODOT ENGINE                                */
+/*                      https://godotengine.org                          */
+/*************************************************************************/
+/* Copyright (c) 2007-2019 Juan Linietsky, Ariel Manzur.                 */
+/* Copyright (c) 2014-2019 Godot Engine contributors (cf. AUTHORS.md)    */
+/*                                                                       */
+/* Permission is hereby granted, free of charge, to any person obtaining */
+/* a copy of this software and associated documentation files (the       */
+/* "Software"), to deal in the Software without restriction, including   */
+/* without limitation the rights to use, copy, modify, merge, publish,   */
+/* distribute, sublicense, and/or sell copies of the Software, and to    */
+/* permit persons to whom the Software is furnished to do so, subject to */
+/* the following conditions:                                             */
+/*                                                                       */
+/* The above copyright notice and this permission notice shall be        */
+/* included in all copies or substantial portions of the Software.       */
+/*                                                                       */
+/* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,       */
+/* EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF    */
+/* MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.*/
+/* IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY  */
+/* CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT,  */
+/* TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION WITH THE     */
+/* SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.                */
+/*************************************************************************/
+
+#ifndef WEBRTC_DATA_CHANNEL_H
+#define WEBRTC_DATA_CHANNEL_H
+
+#include "core/io/packet_peer.h"
+
+class WebRTCDataChannel : public PacketPeer {
+	GDCLASS(WebRTCDataChannel, PacketPeer);
+
+public:
+	enum WriteMode {
+		WRITE_MODE_TEXT,
+		WRITE_MODE_BINARY,
+	};
+
+	enum ChannelState {
+		STATE_CONNECTING,
+		STATE_OPEN,
+		STATE_CLOSING,
+		STATE_CLOSED
+	};
+
+protected:
+	static void _bind_methods();
+
+public:
+	virtual void set_write_mode(WriteMode mode) = 0;
+	virtual WriteMode get_write_mode() const = 0;
+	virtual bool was_string_packet() const = 0;
+
+	virtual ChannelState get_ready_state() const = 0;
+	virtual String get_label() const = 0;
+	virtual bool is_ordered() const = 0;
+	virtual int get_id() const = 0;
+	virtual int get_max_packet_life_time() const = 0;
+	virtual int get_max_retransmits() const = 0;
+	virtual String get_protocol() const = 0;
+	virtual bool is_negotiated() const = 0;
+
+	virtual Error poll() = 0;
+	virtual void close() = 0;
+
+	/** Inherited from PacketPeer: **/
+	virtual int get_available_packet_count() const = 0;
+	virtual Error get_packet(const uint8_t **r_buffer, int &r_buffer_size) = 0; ///< buffer is GONE after next get_packet
+	virtual Error put_packet(const uint8_t *p_buffer, int p_buffer_size) = 0;
+
+	virtual int get_max_packet_size() const = 0;
+
+	WebRTCDataChannel();
+	~WebRTCDataChannel();
+};
+
+VARIANT_ENUM_CAST(WebRTCDataChannel::WriteMode);
+VARIANT_ENUM_CAST(WebRTCDataChannel::ChannelState);
+#endif // WEBRTC_DATA_CHANNEL_H