/**************************************************************************/ /* WebRTCLibPeerConnection.cpp */ /**************************************************************************/ /* This file is part of: */ /* GODOT ENGINE */ /* https://godotengine.org */ /**************************************************************************/ /* Copyright (c) 2014-present Godot Engine contributors (see AUTHORS.md). */ /* Copyright (c) 2007-2014 Juan Linietsky, Ariel Manzur. */ /* */ /* 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 #define OK Error::OK #define FAILED Error::FAILED #define ERR_UNCONFIGURED Error::ERR_UNCONFIGURED #define ERR_INVALID_PARAMETER Error::ERR_INVALID_PARAMETER #endif void WebRTCLibPeerConnection::initialize_signaling() { #ifdef DEBUG_ENABLED rtc::InitLogger(rtc::LogLevel::Debug); #endif } void WebRTCLibPeerConnection::deinitialize_signaling() { rtc::Cleanup(); } 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.maxPacketLifeTime = std::chrono::milliseconds(p_dict["maxPacketLifeTime"].operator int32_t()); } else if (p_dict.has("maxRetransmits")) { r_config.reliability.maxRetransmits = 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; } WebRTCPeerConnection::ConnectionState 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; } } WebRTCLibPeerConnection::GatheringState WebRTCLibPeerConnection::_get_gathering_state() const { ERR_FAIL_COND_V(peer_connection == nullptr, GATHERING_STATE_NEW); rtc::PeerConnection::GatheringState state = peer_connection->gatheringState(); switch (state) { case rtc::PeerConnection::GatheringState::New: return GATHERING_STATE_NEW; case rtc::PeerConnection::GatheringState::InProgress: return GATHERING_STATE_GATHERING; case rtc::PeerConnection::GatheringState::Complete: return GATHERING_STATE_COMPLETE; default: return GATHERING_STATE_NEW; } } WebRTCLibPeerConnection::SignalingState WebRTCLibPeerConnection::_get_signaling_state() const { ERR_FAIL_COND_V(peer_connection == nullptr, SIGNALING_STATE_CLOSED); rtc::PeerConnection::SignalingState state = peer_connection->signalingState(); switch (state) { case rtc::PeerConnection::SignalingState::Stable: return SIGNALING_STATE_STABLE; case rtc::PeerConnection::SignalingState::HaveLocalOffer: return SIGNALING_STATE_HAVE_LOCAL_OFFER; case rtc::PeerConnection::SignalingState::HaveRemoteOffer: return SIGNALING_STATE_HAVE_REMOTE_OFFER; case rtc::PeerConnection::SignalingState::HaveLocalPranswer: return SIGNALING_STATE_HAVE_LOCAL_PRANSWER; case rtc::PeerConnection::SignalingState::HaveRemotePranswer: return SIGNALING_STATE_HAVE_REMOTE_PRANSWER; default: return SIGNALING_STATE_CLOSED; } } Error 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 _create_pc(config); } #if defined(GDNATIVE_WEBRTC) || defined(GDEXTENSION_WEBRTC_40) Object *WebRTCLibPeerConnection::_create_data_channel(const String &p_channel, const Dictionary &p_channel_config) try { #else Ref WebRTCLibPeerConnection::_create_data_channel(const String &p_channel, const Dictionary &p_channel_config) try { #endif 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 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); } Error 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); } Error 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); } Error 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; } #ifdef GDNATIVE_WEBRTC Error WebRTCLibPeerConnection::_add_ice_candidate(const String &sdpMidName, int64_t sdpMlineIndexName, const String &sdpName) try { #else Error WebRTCLibPeerConnection::_add_ice_candidate(const String &sdpMidName, int32_t sdpMlineIndexName, const String &sdpName) try { #endif 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); } Error 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(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 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(); }