// SPDX-FileCopyrightText: 2021 Nheko Contributors // // SPDX-License-Identifier: GPL-3.0-or-later #pragma once #include #include #include #include #include #include #include #include "CallDevices.h" #include "WebRTCSession.h" #include "mtx/events/collections.hpp" #include "mtx/events/voip.hpp" namespace mtx::responses { struct TurnServer; } class QUrl; class CallManager : public QObject { Q_OBJECT Q_PROPERTY(bool haveCallInvite READ haveCallInvite NOTIFY newInviteState) Q_PROPERTY(bool isOnCall READ isOnCall NOTIFY newCallState) Q_PROPERTY(webrtc::CallType callType READ callType NOTIFY newInviteState) Q_PROPERTY(webrtc::State callState READ callState NOTIFY newCallState) Q_PROPERTY(QString callParty READ callParty NOTIFY newInviteState) Q_PROPERTY(QString callPartyDisplayName READ callPartyDisplayName NOTIFY newInviteState) Q_PROPERTY(QString callPartyAvatarUrl READ callPartyAvatarUrl NOTIFY newInviteState) Q_PROPERTY(bool isMicMuted READ isMicMuted NOTIFY micMuteChanged) Q_PROPERTY(bool haveLocalPiP READ haveLocalPiP NOTIFY newCallState) Q_PROPERTY(QStringList mics READ mics NOTIFY devicesChanged) Q_PROPERTY(QStringList cameras READ cameras NOTIFY devicesChanged) Q_PROPERTY(bool callsSupported READ callsSupported CONSTANT) Q_PROPERTY(bool screenShareSupported READ screenShareSupported CONSTANT) public: CallManager(QObject *); bool haveCallInvite() const { return haveCallInvite_; } bool isOnCall() const { return session_.state() != webrtc::State::DISCONNECTED; } webrtc::CallType callType() const { return callType_; } webrtc::State callState() const { return session_.state(); } QString callParty() const { return callParty_; } QString callPartyDisplayName() const { return callPartyDisplayName_; } QString callPartyAvatarUrl() const { return callPartyAvatarUrl_; } bool isMicMuted() const { return session_.isMicMuted(); } bool haveLocalPiP() const { return session_.haveLocalPiP(); } QStringList mics() const { return devices(false); } QStringList cameras() const { return devices(true); } void refreshTurnServer(); static bool callsSupported(); static bool screenShareSupported(); public slots: void sendInvite(const QString &roomid, webrtc::CallType, unsigned int windowIndex = 0); void syncEvent(const mtx::events::collections::TimelineEvents &event); void toggleMicMute(); void toggleLocalPiP() { session_.toggleLocalPiP(); } void acceptInvite(); void hangUp(mtx::events::msg::CallHangUp::Reason = mtx::events::msg::CallHangUp::Reason::User); QStringList windowList(); void previewWindow(unsigned int windowIndex) const; signals: void newMessage(const QString &roomid, const mtx::events::msg::CallInvite &); void newMessage(const QString &roomid, const mtx::events::msg::CallCandidates &); void newMessage(const QString &roomid, const mtx::events::msg::CallAnswer &); void newMessage(const QString &roomid, const mtx::events::msg::CallHangUp &); void newInviteState(); void newCallState(); void micMuteChanged(); void devicesChanged(); void turnServerRetrieved(const mtx::responses::TurnServer &); private slots: void retrieveTurnServer(); private: WebRTCSession &session_; QString roomid_; QString callParty_; QString callPartyDisplayName_; QString callPartyAvatarUrl_; std::string callid_; const uint32_t timeoutms_ = 120000; webrtc::CallType callType_ = webrtc::CallType::VOICE; bool haveCallInvite_ = false; std::string inviteSDP_; std::vector remoteICECandidates_; std::vector turnURIs_; QTimer turnServerTimer_; QMediaPlayer player_; std::vector> windows_; template bool handleEvent(const mtx::events::collections::TimelineEvents &event); void handleEvent(const mtx::events::RoomEvent &); void handleEvent(const mtx::events::RoomEvent &); void handleEvent(const mtx::events::RoomEvent &); void handleEvent(const mtx::events::RoomEvent &); void answerInvite(const mtx::events::msg::CallInvite &, bool isVideo); void generateCallID(); QStringList devices(bool isVideo) const; void clear(); void endCall(); void playRingtone(const QUrl &ringtone, bool repeat); void stopRingtone(); };