// turns protocol events into bridged calls
package calls

import (
	"fmt"
	"sync"
	"time"

	"dev.narayana.im/narayana/telegabber/calls/audio"
	"dev.narayana.im/narayana/telegabber/calls/signaling"
	"dev.narayana.im/narayana/telegabber/calls/signaling/tgsig"
	"dev.narayana.im/narayana/telegabber/calls/signaling/xmppsig"
	"dev.narayana.im/narayana/telegabber/xmpp/jingle"

	"github.com/pion/webrtc/v4"
	log "github.com/sirupsen/logrus"
	"gosrc.io/xmpp/stanza"
)

// look up per-session adapters by bare jid; ok=false drops the call
type SessionLookup func(bareJID string) (tg *tgsig.Adapter, xmpp *xmppsig.Adapter, ok bool)

type Config struct {
	JingleManager *jingle.Manager
	TgManager     *tgsig.Manager
	Lookup        SessionLookup

	// zero falls through to signaling defaults (60s/20s/15s)
	RingTimeout     time.Duration
	ExchangeTimeout time.Duration
	ConnectTimeout  time.Duration
}

type Orchestrator struct {
	cfg Config
}

func New(cfg Config) *Orchestrator {
	o := &Orchestrator{cfg: cfg}
	if cfg.JingleManager != nil {
		cfg.JingleManager.OnProposal = o.onProposal
	}
	return o
}

// per-side surfaces the audio layer needs
type tgAudioSide interface {
	NtgClient() audio.NtgClient
	ChatID() int64
}

type xmppAudioSide interface {
	PeerConnection() *webrtc.PeerConnection
	// swap in the orchestrator's track handler and drain buffered tracks
	SetTrackHandler(fn func(t *webrtc.TrackRemote))
}

type Call struct {
	bridge *signaling.Bridge

	mu            sync.Mutex
	tgSide        tgAudioSide
	xmppSide      xmppAudioSide
	audioTgToXmpp *audio.TgToXmpp
	audioXmppToTg *audio.XmppToTg

	closeOnce sync.Once
}

func (c *Call) Bridge() *signaling.Bridge { return c.bridge }

// build the audio halves once the negotiated opus fmtp is in the SDP
func (c *Call) setupAudio(remoteSDP string) {
	remoteFmtp := audio.ExtractOpusFmtp(remoteSDP)
	c.mu.Lock()
	if c.audioTgToXmpp != nil || c.audioXmppToTg != nil {
		c.mu.Unlock()
		return
	}
	tgSide, xmppSide := c.tgSide, c.xmppSide
	c.mu.Unlock()
	if tgSide == nil || xmppSide == nil {
		return
	}

	ntg := tgSide.NtgClient()
	pc := xmppSide.PeerConnection()
	if ntg == nil || pc == nil {
		// signaling-only test setup
		return
	}
	chatID := tgSide.ChatID()

	tgToXmpp, err := audio.NewTgToXmpp(audio.TgToXmppOptions{
		Ntg:        ntg,
		ChatID:     chatID,
		RemoteFmtp: remoteFmtp,
	})
	if err != nil {
		log.WithError(err).Error("setupAudio: NewTgToXmpp")
		return
	}
	xmppToTg, err := audio.NewXmppToTg(audio.XmppToTgOptions{
		Ntg:    ntg,
		ChatID: chatID,
	})
	if err != nil {
		_ = tgToXmpp.Close()
		log.WithError(err).Error("setupAudio: NewXmppToTg")
		return
	}

	if err := attachOutgoingTrack(pc, tgToXmpp.LocalTrack()); err != nil {
		log.WithField("chat_id", chatID).WithError(err).Warn("setupAudio: attach outgoing track failed")
	}
	xmppSide.SetTrackHandler(xmppToTg.HandleTrack)
	log.WithFields(log.Fields{
		"chat_id":     chatID,
		"remote_fmtp": remoteFmtp,
	}).Info("setupAudio: halves wired; awaiting startAudio on bridge OnEstablished")

	c.mu.Lock()
	c.audioTgToXmpp = tgToXmpp
	c.audioXmppToTg = xmppToTg
	c.mu.Unlock()
}

// swap the real outgoing track into the placeholder sender from buildPC
func attachOutgoingTrack(pc *webrtc.PeerConnection, track *webrtc.TrackLocalStaticSample) error {
	for _, s := range pc.GetSenders() {
		if s == nil {
			continue
		}
		if err := s.ReplaceTrack(track); err == nil {
			return nil
		}
	}
	return fmt.Errorf("no sender accepted track")
}

// fired by Bridge.OnEstablished
func (c *Call) startAudio() {
	c.mu.Lock()
	tgHalf := c.audioTgToXmpp
	xmppHalf := c.audioXmppToTg
	c.mu.Unlock()
	if tgHalf == nil || xmppHalf == nil {
		return
	}
	if err := tgHalf.Start(); err != nil {
		log.WithError(err).Warn("tg->xmpp audio Start failed")
	}
	if err := xmppHalf.Start(); err != nil {
		log.WithError(err).Warn("xmpp->tg audio Start failed")
	}
}

func (c *Call) Close() {
	c.closeOnce.Do(func() {
		c.mu.Lock()
		tgHalf := c.audioTgToXmpp
		xmppHalf := c.audioXmppToTg
		tgSide := c.tgSide
		c.audioTgToXmpp = nil
		c.audioXmppToTg = nil
		c.mu.Unlock()
		if tgHalf != nil {
			_ = tgHalf.Close()
		}
		if xmppHalf != nil {
			_ = xmppHalf.Close()
		}
		// release ntg streams set in setupAudio; both halves are built
		// together, skipped in signaling-only test setups
		if (tgHalf != nil || xmppHalf != nil) && tgSide != nil {
			if ntg := tgSide.NtgClient(); ntg != nil {
				_ = ntg.ClearStreams(tgSide.ChatID())
			}
		}
	})
}

// inbound JMI <propose>; postReady fires after the session is in StateRinging
// so a fast tdlib Pending{IsReceived:true} doesn't drop the JMI <ringing/>
func (o *Orchestrator) onProposal(p jingle.IncomingProposal) (*jingle.Session, func(), error) {
	j, err := stanza.NewJid(p.From)
	if err != nil {
		return nil, nil, nil
	}
	tg, xmpp, ok := o.cfg.Lookup(j.Bare())
	if !ok || tg == nil || xmpp == nil {
		return nil, nil, nil
	}
	userID, ok := xmppsig.ParseTargetJID(p.To)
	if !ok {
		return nil, nil, nil
	}

	callee, sess, err := xmpp.NewCallee(p)
	if err != nil {
		return nil, nil, err
	}
	caller := tg.NewCaller(userID, false) // audio only

	call := &Call{tgSide: caller, xmppSide: callee}
	br := o.newBridge(caller, callee, call)
	call.bridge = br
	callee.Bind(br)
	caller.Bind(br)

	// defer audio half construction until SDP lands so a pre-SDP cancel
	// doesn't spin up ntgcalls
	callee.SetOnRemoteSDP(call.setupAudio)

	// off the dispatcher goroutine - Caller.Start hits tdlib CreateCall
	return sess, func() { go br.Start() }, nil
}

// IncomingCallHandler for tgsig.Manager
func (o *Orchestrator) NewFromTelegram(jid string, callID int32, userID int64, isVideo bool) (ok bool) {
	entry := log.WithFields(log.Fields{
		"jid":      jid,
		"call_id":  callID,
		"user_id":  userID,
		"is_video": isVideo,
	})
	defer func() {
		if r := recover(); r != nil {
			entry.WithField("panic", r).Error("NewFromTelegram: panic recovered")
			ok = false
		}
	}()

	tg, xmpp, lookupOk := o.cfg.Lookup(jid)
	if !lookupOk || tg == nil || xmpp == nil {
		entry.Warn("NewFromTelegram: no live session for jid")
		return false
	}

	// xmpp side first - it's the only fallible step, otherwise a failure here
	// would leak the tg side until tdlib timed the call out
	caller, err := xmpp.NewCaller(jid, userID)
	if err != nil {
		entry.WithError(err).Error("NewFromTelegram: NewCaller failed")
		return false
	}
	callee := tg.NewCallee(callID, userID, isVideo)

	call := &Call{tgSide: callee, xmppSide: caller}
	br := o.newBridge(caller, callee, call)
	call.bridge = br
	callee.Bind(br)
	caller.Bind(br)

	caller.SetOnRemoteSDP(call.setupAudio)
	br.Start()
	return true
}

func (o *Orchestrator) newBridge(caller signaling.Caller, callee signaling.Callee, call *Call) *signaling.Bridge {
	timers := signaling.NewStdTimers()
	br := signaling.New(signaling.Config{
		Caller:          caller,
		Callee:          callee,
		Timers:          timers,
		RingTimeout:     o.cfg.RingTimeout,
		ExchangeTimeout: o.cfg.ExchangeTimeout,
		ConnectTimeout:  o.cfg.ConnectTimeout,
		OnEstablished:   call.startAudio,
		OnTerminated:    call.Close,
	})
	timers.SetBridge(br)
	return br
}