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/*************************************************************************/
/* WebRTCLibPeerConnection.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 "WebRTCLibPeerConnection.hpp"
#include "WebRTCLibDataChannel.hpp"
using namespace godot;
using namespace godot_webrtc;
#ifdef GDNATIVE_WEBRTC
struct CastableError {
godot::Error err_enum;
int64_t err_int;
operator int64_t() { return err_int; }
operator godot::Error() { return err_enum; }
CastableError(godot::Error p_enum, int64_t p_int) {
err_enum = p_enum;
err_int = p_int;
}
};
#define MKERR(m_err) CastableError(godot::Error::m_err, GODOT_##m_err)
#define OK MKERR(OK)
#define FAILED MKERR(FAILED)
#define ERR_UNCONFIGURED MKERR(ERR_UNCONFIGURED)
#define ERR_UNAVAILABLE MKERR(ERR_UNAVAILABLE)
#define ERR_INVALID_PARAMETER MKERR(ERR_INVALID_PARAMETER)
#define ERR_BUG MKERR(ERR_BUG)
#endif
void WebRTCLibPeerConnection::initialize_signaling() {
#ifdef DEBUG_ENABLED
rtc::InitLogger(rtc::LogLevel::Debug);
#endif
}
void WebRTCLibPeerConnection::deinitialize_signaling() {
}
Error WebRTCLibPeerConnection::_parse_ice_server(rtc::Configuration &r_config, Dictionary p_server) {
ERR_FAIL_COND_V(!p_server.has("urls"), ERR_INVALID_PARAMETER);
// Parse mandatory URL
Array urls;
Variant urls_var = p_server["urls"];
if (urls_var.get_type() == Variant::STRING) {
urls.push_back(urls_var);
} else if (urls_var.get_type() == Variant::ARRAY) {
urls = urls_var;
} else {
ERR_FAIL_V(ERR_INVALID_PARAMETER);
}
// Parse credentials (only meaningful for TURN, only support password)
String username;
String credential;
if (p_server.has("username") && p_server["username"].get_type() == Variant::STRING) {
username = p_server["username"];
}
if (p_server.has("credential") && p_server["credential"].get_type() == Variant::STRING) {
credential = p_server["credential"];
}
for (int i = 0; i < urls.size(); i++) {
rtc::IceServer srv(urls[i].operator String().utf8().get_data());
srv.username = username.utf8().get_data();
srv.password = credential.utf8().get_data();
r_config.iceServers.push_back(srv);
}
return OK;
}
Error WebRTCLibPeerConnection::_parse_channel_config(rtc::DataChannelInit &r_config, const Dictionary &p_dict) {
Variant nil;
Variant v;
if (p_dict.has("negotiated")) {
r_config.negotiated = p_dict["negotiated"].operator bool();
}
if (p_dict.has("id")) {
r_config.id = uint16_t(p_dict["id"].operator int32_t());
}
// If negotiated it must have an ID, and ID only makes sense when negotiated.
ERR_FAIL_COND_V(r_config.negotiated != r_config.id.has_value(), ERR_INVALID_PARAMETER);
// Channels cannot be both time-constrained and retry-constrained.
ERR_FAIL_COND_V(p_dict.has("maxPacketLifeTime") && p_dict.has("maxRetransmits"), ERR_INVALID_PARAMETER);
if (p_dict.has("maxPacketLifeTime")) {
r_config.reliability.type = rtc::Reliability::Type::Timed;
r_config.reliability.rexmit = std::chrono::milliseconds(p_dict["maxPacketLifeTime"].operator int32_t());
} else if (p_dict.has("maxRetransmits")) {
r_config.reliability.type = rtc::Reliability::Type::Rexmit;
r_config.reliability.rexmit = p_dict["maxRetransmits"].operator int32_t();
}
if (p_dict.has("ordered") && p_dict["ordered"].operator bool() == false) {
r_config.reliability.unordered = true;
}
if (p_dict.has("protocol")) {
r_config.protocol = p_dict["protocol"].operator String().utf8().get_data();
}
return OK;
}
int64_t WebRTCLibPeerConnection::_get_connection_state() const {
ERR_FAIL_COND_V(peer_connection == nullptr, STATE_CLOSED);
rtc::PeerConnection::State state = peer_connection->state();
switch (state) {
case rtc::PeerConnection::State::New:
return STATE_NEW;
case rtc::PeerConnection::State::Connecting:
return STATE_CONNECTING;
case rtc::PeerConnection::State::Connected:
return STATE_CONNECTED;
case rtc::PeerConnection::State::Disconnected:
return STATE_DISCONNECTED;
case rtc::PeerConnection::State::Failed:
return STATE_FAILED;
default:
return STATE_CLOSED;
}
}
int64_t WebRTCLibPeerConnection::_initialize(const Dictionary &p_config) {
rtc::Configuration config = {};
if (p_config.has("iceServers") && p_config["iceServers"].get_type() == Variant::ARRAY) {
Array servers = p_config["iceServers"];
for (int i = 0; i < servers.size(); i++) {
ERR_FAIL_COND_V(servers[i].get_type() != Variant::DICTIONARY, ERR_INVALID_PARAMETER);
Dictionary server = servers[i];
Error err = _parse_ice_server(config, server);
ERR_FAIL_COND_V(err != OK, FAILED);
}
}
return (int64_t)_create_pc(config);
}
Object *WebRTCLibPeerConnection::_create_data_channel(const String &p_channel, const Dictionary &p_channel_config) try {
ERR_FAIL_COND_V(!peer_connection, nullptr);
// Read config from dictionary
rtc::DataChannelInit config;
Error err = _parse_channel_config(config, p_channel_config);
ERR_FAIL_COND_V(err != OK, nullptr);
std::shared_ptr<rtc::DataChannel> ch = peer_connection->createDataChannel(p_channel.utf8().get_data(), config);
ERR_FAIL_COND_V(ch == nullptr, nullptr);
WebRTCLibDataChannel *wrapper = WebRTCLibDataChannel::new_data_channel(ch, ch->id().has_value());
ERR_FAIL_COND_V(wrapper == nullptr, nullptr);
return wrapper;
} catch (const std::exception &e) {
ERR_PRINT(e.what());
ERR_FAIL_V(nullptr);
}
int64_t WebRTCLibPeerConnection::_create_offer() try {
ERR_FAIL_COND_V(!peer_connection, ERR_UNCONFIGURED);
ERR_FAIL_COND_V(_get_connection_state() != STATE_NEW, FAILED);
peer_connection->setLocalDescription(rtc::Description::Type::Offer);
return OK;
} catch (const std::exception &e) {
ERR_PRINT(e.what());
ERR_FAIL_V(FAILED);
}
int64_t WebRTCLibPeerConnection::_set_remote_description(const String &p_type, const String &p_sdp) try {
ERR_FAIL_COND_V(!peer_connection, ERR_UNCONFIGURED);
std::string sdp(p_sdp.utf8().get_data());
std::string type(p_type.utf8().get_data());
rtc::Description desc(sdp, type);
peer_connection->setRemoteDescription(desc);
// Automatically create the answer.
if (p_type == String("offer")) {
peer_connection->setLocalDescription(rtc::Description::Type::Answer);
}
return OK;
} catch (const std::exception &e) {
ERR_PRINT(e.what());
ERR_FAIL_V(FAILED);
}
int64_t WebRTCLibPeerConnection::_set_local_description(const String &p_type, const String &p_sdp) {
ERR_FAIL_COND_V(!peer_connection, ERR_UNCONFIGURED);
// XXX Library quirk. It doesn't seem possible to create offers/answers without setting the local description.
// Ignore this call for now to avoid crash (it's already set automatically!).
//peer_connection->setLocalDescription(p_type == String("offer") ? rtc::Description::Type::Offer : rtc::Description::Type::Answer);
return OK;
}
int64_t WebRTCLibPeerConnection::_add_ice_candidate(const String &sdpMidName, int64_t sdpMlineIndexName, const String &sdpName) try {
ERR_FAIL_COND_V(!peer_connection, ERR_UNCONFIGURED);
rtc::Candidate candidate(sdpName.utf8().get_data(), sdpMidName.utf8().get_data());
peer_connection->addRemoteCandidate(candidate);
return OK;
} catch (const std::exception &e) {
ERR_PRINT(e.what());
ERR_FAIL_V(FAILED);
}
int64_t WebRTCLibPeerConnection::_poll() {
ERR_FAIL_COND_V(!peer_connection, ERR_UNCONFIGURED);
while (!signal_queue.empty()) {
mutex_signal_queue->lock();
Signal signal = signal_queue.front();
signal_queue.pop();
mutex_signal_queue->unlock();
signal.emit(this);
}
return OK;
}
void WebRTCLibPeerConnection::_close() {
if (peer_connection != nullptr) {
try {
peer_connection->close();
} catch (...) {
}
}
while (!signal_queue.empty()) {
signal_queue.pop();
}
}
void WebRTCLibPeerConnection::_init() {
#ifdef GDNATIVE_WEBRTC
register_interface(&interface);
#endif
mutex_signal_queue = new std::mutex;
_initialize(Dictionary());
}
Error WebRTCLibPeerConnection::_create_pc(rtc::Configuration &r_config) try {
// Prevents libdatachannel from automatically creating offers.
r_config.disableAutoNegotiation = true;
peer_connection = std::make_shared<rtc::PeerConnection>(r_config);
ERR_FAIL_COND_V(!peer_connection, FAILED);
// Binding this should be fine as long as we call close when going out of scope.
peer_connection->onLocalDescription([this](rtc::Description description) {
String type = description.type() == rtc::Description::Type::Offer ? "offer" : "answer";
queue_signal("session_description_created", 2, type, String(std::string(description).c_str()));
});
peer_connection->onLocalCandidate([this](rtc::Candidate candidate) {
queue_signal("ice_candidate_created", 3, String(candidate.mid().c_str()), 0, String(candidate.candidate().c_str()));
});
peer_connection->onDataChannel([this](std::shared_ptr<rtc::DataChannel> channel) {
queue_signal("data_channel_received", 1, WebRTCLibDataChannel::new_data_channel(channel, false));
});
/*
peer_connection->onStateChange([](rtc::PeerConnection::State state) {
std::cout << "[State: " << state << "]" << std::endl;
});
peer_connection->onGatheringStateChange([](rtc::PeerConnection::GatheringState state) {
std::cout << "[Gathering State: " << state << "]" << std::endl;
});
*/
return OK;
} catch (const std::exception &e) {
ERR_PRINT(e.what());
ERR_FAIL_V(FAILED);
}
WebRTCLibPeerConnection::WebRTCLibPeerConnection() {
#ifndef GDNATIVE_WEBRTC
_init();
#endif
}
WebRTCLibPeerConnection::~WebRTCLibPeerConnection() {
#ifdef GDNATIVE_WEBRTC
if (_owner) {
register_interface(nullptr);
}
#endif
_close();
delete mutex_signal_queue;
}
void WebRTCLibPeerConnection::queue_signal(String p_name, int p_argc, const Variant &p_arg1, const Variant &p_arg2, const Variant &p_arg3) {
mutex_signal_queue->lock();
const Variant argv[3] = { p_arg1, p_arg2, p_arg3 };
signal_queue.push(Signal(p_name, p_argc, argv));
mutex_signal_queue->unlock();
}
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