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/*************************************************************************/
/* WebRTCLibDataChannel.cpp */
/*************************************************************************/
/* This file is part of: */
/* GODOT ENGINE */
/* https://godotengine.org */
/*************************************************************************/
/* Copyright (c) 2007-2021 Juan Linietsky, Ariel Manzur. */
/* Copyright (c) 2014-2021 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 "WebRTCLibDataChannel.hpp"
#ifdef GDNATIVE_WEBRTC
#include "GDNativeLibrary.hpp"
#include "NativeScript.hpp"
#define ERR_UNAVAILABLE Error::ERR_UNAVAILABLE
#define FAILED Error::FAILED
#define ERR_INVALID_PARAMETER Error::ERR_INVALID_PARAMETER
#define OK Error::OK
#endif
#include <stdio.h>
#include <string.h>
#include <cstring>
using namespace godot;
using namespace godot_webrtc;
// DataChannel
WebRTCLibDataChannel *WebRTCLibDataChannel::new_data_channel(std::shared_ptr<rtc::DataChannel> p_channel, bool p_negotiated) {
// Invalid channel result in NULL return
ERR_FAIL_COND_V(!p_channel, nullptr);
#ifdef GDNATIVE_WEBRTC
// Instance a WebRTCDataChannelGDNative object
WebRTCDataChannelGDNative *native = WebRTCDataChannelGDNative::_new();
// Set our implementation as its script
NativeScript *script = NativeScript::_new();
script->set_library(detail::get_wrapper<GDNativeLibrary>((godot_object *)gdnlib));
script->set_class_name("WebRTCLibDataChannel");
native->set_script(script);
WebRTCLibDataChannel *out = native->cast_to<WebRTCLibDataChannel>(native);
#else
WebRTCLibDataChannel *out = memnew(WebRTCLibDataChannel);
#endif
// Bind the library data channel to our object.
out->bind_channel(p_channel, p_negotiated);
return out;
}
void WebRTCLibDataChannel::bind_channel(std::shared_ptr<rtc::DataChannel> p_channel, bool p_negotiated) {
ERR_FAIL_COND(!p_channel);
channel = p_channel;
negotiated = p_negotiated;
// Binding this should be fine as long as we call close when going out of scope.
p_channel->onMessage([this](auto message) {
if (std::holds_alternative<rtc::string>(message)) {
rtc::string str = std::get<rtc::string>(message);
queue_packet(reinterpret_cast<const uint8_t *>(str.c_str()), str.size(), true);
} else if (std::holds_alternative<rtc::binary>(message)) {
rtc::binary bin = std::get<rtc::binary>(message);
queue_packet(reinterpret_cast<const uint8_t *>(&bin[0]), bin.size(), false);
} else {
ERR_PRINT("Message parsing bug. Unknown message type.");
}
});
p_channel->onOpen([this]() {
channel_state = STATE_OPEN;
});
p_channel->onClosed([this]() {
channel_state = STATE_CLOSED;
});
p_channel->onError([](auto error) {
ERR_PRINT("Channel Error: " + String(std::string(error).c_str()));
});
}
void WebRTCLibDataChannel::queue_packet(const uint8_t *data, uint32_t size, bool p_is_string) {
mutex->lock();
std::vector<uint8_t> packet;
packet.resize(size);
memcpy(&packet[0], data, size);
packet_queue.push(QueuedPacket(packet, p_is_string));
mutex->unlock();
}
void WebRTCLibDataChannel::_set_write_mode(WriteMode p_mode) {
ERR_FAIL_COND(p_mode != WRITE_MODE_TEXT && p_mode != WRITE_MODE_BINARY);
write_mode = p_mode;
}
WebRTCDataChannel::WriteMode WebRTCLibDataChannel::_get_write_mode() const {
return write_mode;
}
bool WebRTCLibDataChannel::_was_string_packet() const {
return current_packet.second;
}
WebRTCDataChannel::ChannelState WebRTCLibDataChannel::_get_ready_state() const {
ERR_FAIL_COND_V(!channel, STATE_CLOSED);
return channel_state;
}
String WebRTCLibDataChannel::_get_label() const {
ERR_FAIL_COND_V(!channel, "");
return channel->label().c_str();
}
bool WebRTCLibDataChannel::_is_ordered() const {
ERR_FAIL_COND_V(!channel, false);
return channel->reliability().unordered == false;
}
int64_t WebRTCLibDataChannel::_get_id() const {
ERR_FAIL_COND_V(!channel, -1);
return channel->id().value_or(-1);
}
int64_t WebRTCLibDataChannel::_get_max_packet_life_time() const {
ERR_FAIL_COND_V(!channel, 0);
return channel->reliability().type == rtc::Reliability::Type::Timed ? std::get<std::chrono::milliseconds>(channel->reliability().rexmit).count() : -1;
}
int64_t WebRTCLibDataChannel::_get_max_retransmits() const {
ERR_FAIL_COND_V(!channel, 0);
return channel->reliability().type == rtc::Reliability::Type::Rexmit ? std::get<int>(channel->reliability().rexmit) : -1;
}
String WebRTCLibDataChannel::_get_protocol() const {
ERR_FAIL_COND_V(!channel, "");
return channel->protocol().c_str();
}
bool WebRTCLibDataChannel::_is_negotiated() const {
ERR_FAIL_COND_V(!channel, false);
return negotiated;
}
int64_t WebRTCLibDataChannel::_get_buffered_amount() const {
ERR_FAIL_COND_V(!channel, 0);
return channel->bufferedAmount();
}
Error WebRTCLibDataChannel::_poll() {
return OK;
}
void WebRTCLibDataChannel::_close() try {
if (channel) {
channel->close();
}
} catch (...) {
}
Error WebRTCLibDataChannel::_get_packet(const uint8_t **r_buffer, int32_t *r_len) {
ERR_FAIL_COND_V(packet_queue.empty(), ERR_UNAVAILABLE);
mutex->lock();
// Update current packet and pop queue
current_packet = packet_queue.front();
packet_queue.pop();
// Set out buffer and size (buffer will be gone at next get_packet or close)
*r_buffer = ¤t_packet.first[0];
*r_len = current_packet.first.size();
mutex->unlock();
return OK;
}
Error WebRTCLibDataChannel::_put_packet(const uint8_t *p_buffer, int64_t p_len) try {
ERR_FAIL_COND_V(!channel, FAILED);
ERR_FAIL_COND_V(channel->isClosed(), FAILED);
if (write_mode == WRITE_MODE_TEXT) {
std::string str(p_len, '\x00');
std::strncpy(str.data(), (const char *)p_buffer, p_len);
channel->send(str);
} else if (write_mode == WRITE_MODE_BINARY) {
channel->send(reinterpret_cast<const std::byte *>(p_buffer), p_len);
} else {
ERR_FAIL_V(ERR_INVALID_PARAMETER);
}
return OK;
} catch (const std::exception &e) {
ERR_PRINT(e.what());
ERR_FAIL_V(FAILED);
}
int64_t WebRTCLibDataChannel::_get_available_packet_count() const {
return packet_queue.size();
}
int64_t WebRTCLibDataChannel::_get_max_packet_size() const {
return 16384; // See RFC-8831 section 6.6: https://datatracker.ietf.org/doc/rfc8831/
}
WebRTCLibDataChannel::WebRTCLibDataChannel() {
mutex = new std::mutex;
}
WebRTCLibDataChannel::~WebRTCLibDataChannel() {
_close();
channel = nullptr;
delete mutex;
}
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