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/*

 * Copyright (C) 2022 by Andreas Theofilu <andreas@theosys.at>

 *

 * This program is free software; you can redistribute it and/or modify

 * it under the terms of the GNU General Public License as published by

 * the Free Software Foundation; either version 3 of the License, or

 * (at your option) any later version.

 *

 * This program is distributed in the hope that it will be useful,

 * but WITHOUT ANY WARRANTY; without even the implied warranty of

 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the

 * GNU General Public License for more details.

 *

 * You should have received a copy of the GNU General Public License

 * along with this program; if not, write to the Free Software Foundation,

 * Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA

 */

#ifndef __TSIPCLIENT_H__

#define __TSIPCLIENT_H__

#include <thread>

#include <atomic>

#include <pjsua-lib/pjsua.h>

#ifndef _NOSIP_

#define SIP_MAX_LINES       2

#define current_acc         pjsua_acc_get_default()

#define PJSUA_APP_NO_LIMIT_DURATION (int)0x7FFFFFFF

// Ringtones                US         UK

#define RINGBACK_FREQ1      440     // 400

#define RINGBACK_FREQ2      480     // 450

#define RINGBACK_ON         2000    // 400

#define RINGBACK_OFF        4000    // 200

#define RINGBACK_CNT        1       // 2

#define RINGBACK_INTERVAL   4000    // 2000

#define RING_FREQ1          800

#define RING_FREQ2          640

#define RING_ON             200

#define RING_OFF            100

#define RING_CNT            3

#define RING_INTERVAL       3000

typedef unsigned int        uint_t;

class TSIPClient

{

    public:

        typedef enum SIP_STATE_t

        {

            SIP_NONE,               // Undefined state (only valid on startup before initialisation)

            SIP_IDLE,               // Initialized but no event

            SIP_CONNECTED,          // Call is in progress

            SIP_DISCONNECTED,       // Call has ended

            SIP_TRYING,             // Trying to call someone

            SIP_RINGING,            // Phone is ringing (incoming call)

            SIP_HOLD,               // Active call is paused

            SIP_REJECTED,           // Outgoing call was rejected

            SIP_ERROR               // An error occured

        }SIP_STATE_t;

        TSIPClient();

        ~TSIPClient();

        bool init();

        void cleanUp();

        int getLineID() { return mLine; }

        SIP_STATE_t getSIPState(pjsua_call_id id) { if (id >= 0 && id < PJSUA_MAX_CALLS) return mSIPState[id]; else return SIP_NONE; }

        bool isRegistered() { return mRegistered; }

        static TSIPClient *getSIPClient() { return mMyself; }

        bool call(const std::string& dest);         //<! Start a phone call

        bool pickup(pjsua_call_id call);            //<! Lift up if the phone is ringing

        bool terminate(int id);                     //<! Terminate a call

        bool hold(int id);                          //<! Pause a call

        bool resume(int id);                        //<! Resume a paused call

        bool sendDTMF(std::string& dtmf);           //<! Send a DTMF string

        bool sendLinestate();                       //<! Queries the state of each of the connections used by the SIP device.

        bool sendPrivate(bool state);               //<! Enables or disables the privacy feature on the phone (do not disturb).

        bool redial();                              //<! Redial last number

        bool transfer(int id, const std::string& num);  //<! transfer the current call

        bool setDTMFduration(uint_t ms);            //<! Set the DTMF duration in ms.

        bool getPrivate() { return mDoNotDisturb; } //<! Returns the current private mode.

    protected:

        void runRinger();                           //<! Plays a ring tone if a call is coming.

        static void _log_call(int level, const char *data, int len);

        static void on_incoming_call(pjsua_acc_id acc_id, pjsua_call_id call_id, pjsip_rx_data *rdata);

        static void on_call_state(pjsua_call_id call_id, pjsip_event *e);

        static void on_call_media_state(pjsua_call_id call_id);

        static void call_on_dtmf_callback2(pjsua_call_id call_id, const pjsua_dtmf_info *info);

        static pjsip_redirect_op call_on_redirected(pjsua_call_id call_id, const pjsip_uri *target, const pjsip_event *e);

        static void on_call_transfer_status(pjsua_call_id call_id, int status_code, const pj_str_t *status_text, pj_bool_t final, pj_bool_t *p_cont);

        static void on_transport_state(pjsip_transport *tp, pjsip_transport_state state, const pjsip_transport_state_info *info);

        static void on_ip_change_progress(pjsua_ip_change_op op, pj_status_t status, const pjsua_ip_change_op_info *info);

        static void simple_registrar(pjsip_rx_data *rdata);

        static pj_bool_t default_mod_on_rx_request(pjsip_rx_data *rdata);

        static void ringback_start(pjsua_call_id call_id);

        static void ring_start(pjsua_call_id call_id);

        static void ring_stop(pjsua_call_id call_id);

        static pj_bool_t find_next_call(void);

        static void call_timeout_callback(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry);

        static void on_playfile_done(pjmedia_port *port, void *usr_data);

        static void hangup_timeout_callback(pj_timer_heap_t *timer_heap, struct pj_timer_entry *entry);

    private:

        static pjsip_module mod_default_handler;

        int getNumberCalls();

        pjsua_call_id getActiveCall();

        void setSIPState(SIP_STATE_t s, pjsua_call_id id) { if (id >= 0 && id < PJSUA_MAX_CALLS) mSIPState[id] = s; }

        static void sendConnectionStatus(SIP_STATE_t state, int id);

        static void init_ringtone_player();

        static pj_status_t start_ring_tone();

        static pj_status_t stop_ring_tone();

        int mLine{0};

        bool mRegistered{false};

        SIP_STATE_t mSIPState[PJSUA_MAX_CALLS];

        pjsua_acc_id mAccountID{0};

        std::string mLastCall;

        uint_t mDTMFduration{PJSUA_CALL_SEND_DTMF_DURATION_DEFAULT};

        bool mDoNotDisturb{false};

        static TSIPClient *mMyself;

        static pjsua_call_id mCurrentCall;

        static std::atomic<bool> mRefreshRun;

        static bool mPhoneRingInit;

        // Here is the configuration for the PJSUA library

        typedef struct app_call_data

        {

            pj_timer_entry      timer;

            pj_bool_t           ringback_on{PJ_FALSE};

            pj_bool_t           ring_on{PJ_FALSE};

        } app_call_data;

        typedef struct ringtone_port_info_t

        {

            pj_bool_t ring_on{PJ_FALSE};

            int ring_slot{0};

            pjmedia_port *ring_port{nullptr};

            pj_pool_t *pool{nullptr};

        } ringtone_port_info_t;

        typedef struct pjsua_app_config

        {

            pjsua_config        cfg;

            pjsua_logging_config log_cfg;

            pjsua_media_config  media_cfg;

            pj_bool_t           no_refersub{PJ_FALSE};

            pj_bool_t           enable_qos{PJ_FALSE};

            pj_bool_t           no_tcp{PJ_TRUE};

            pj_bool_t           no_udp{PJ_FALSE};

            pj_bool_t           use_tls{PJ_FALSE};

            pjsua_transport_config  udp_cfg;

            pjsua_transport_config  rtp_cfg;

            pjsip_redirect_op   redir_op;

            int                 srtp_keying{0};

            uint_t              acc_cnt{0};

            pjsua_acc_config    acc_cfg[PJSUA_MAX_ACC];

            uint_t              buddy_cnt{0};

            pjsua_buddy_config  buddy_cfg[PJSUA_MAX_BUDDIES];

            app_call_data       call_data[PJSUA_MAX_CALLS];

            pj_pool_t          *pool{nullptr};

            /* Compatibility with older pjsua */

            uint_t              codec_cnt{0};

            pj_str_t            codec_arg[32];

            uint_t              codec_dis_cnt{0};

            pj_str_t            codec_dis[32];

            pj_bool_t           null_audio{PJ_FALSE};

            uint_t              wav_count{0};

            pj_str_t            wav_files[32];

            uint_t              tone_count{0};

            pjmedia_tone_desc   tones[32];

            pjsua_conf_port_id  tone_slots[32];

            pjsua_player_id     wav_id{0};

            pjsua_conf_port_id  wav_port{0};

            pj_bool_t           auto_play{PJ_FALSE};

            pj_bool_t           auto_play_hangup{PJ_FALSE};

            pj_timer_entry      auto_hangup_timer;

            pj_bool_t           auto_loop{PJ_FALSE};

            pj_bool_t           auto_conf{PJ_FALSE};

            pj_str_t            rec_file;

            pj_bool_t           auto_rec{PJ_FALSE};

            pjsua_recorder_id   rec_id{0};

            pjsua_conf_port_id  rec_port{0};

            unsigned            auto_answer{0};

            unsigned            duration{36000};

            float               mic_level{100.0};

            float               speaker_level{100.0};

            int                 capture_dev{0};

            int                 playback_dev{0};

            uint_t              capture_lat{0};

            uint_t              playback_lat{0};

            pj_bool_t           no_tones{PJ_FALSE};

            int                 ringback_slot{0};

            int                 ringback_cnt{0};

            pjmedia_port        *ringback_port{nullptr};

            int                 ring_slot{0};

            int                 ring_cnt{0};

            pjmedia_port        *ring_port{nullptr};

            uint_t              aud_cnt{0};

        }pjsua_app_config;

        static pjsua_app_config mAppConfig;

        static ringtone_port_info_t mRingtonePortInfo;

};

#endif  // _NOSIP_

#endif  // __TSIPCLIENT_H__