/**
 * Praxis voice session hook — wraps the Pipecat client + SmallWebRTCTransport.
 *
 * Connects to the server's POST /pipecat/webrtc endpoint, manages mic permission,
 * audio playback, live transcript, and a latency readout (ASR→TTS-first-audio).
 *
 * v0.1 SLICE-02: minimal start/speak/reply loop. SLICE-05 expands to the full
 * start → live → debrief session flow.
 */
import { useCallback, useEffect, useRef, useState } from 'react'
import { PipecatClient, type PipecatClientOptions } from '@pipecat-ai/client-js'
import { SmallWebRTCTransport } from '@pipecat-ai/small-webrtc-transport'

export type SessionState = 'idle' | 'connecting' | 'connected' | 'error'

export interface TranscriptEntry {
  role: 'user' | 'assistant'
  text: string
  ts: number
}

export interface LatencyReading {
  /** ms from bot-ready to first assistant audio (approx ASR→TTS first audio). */
  e2eMs: number | null
  label: string
}

export interface UseVoiceSessionResult {
  state: SessionState
  error: string | null
  transcripts: TranscriptEntry[]
  latency: LatencyReading | null
  start: () => Promise<void>
  stop: () => Promise<void>
}

const SERVER_OFFER_URL = '/pipecat/webrtc'

export function useVoiceSession(): UseVoiceSessionResult {
  const [state, setState] = useState<SessionState>('idle')
  const [error, setError] = useState<string | null>(null)
  const [transcripts, setTranscripts] = useState<TranscriptEntry[]>([])
  const [latency, setLatency] = useState<LatencyReading | null>(null)
  const clientRef = useRef<PipecatClient | null>(null)
  const readyAtRef = useRef<number | null>(null)

  const stop = useCallback(async () => {
    const c = clientRef.current
    if (c) {
      try {
        await c.disconnect()
      } catch {
        /* ignore */
      }
      clientRef.current = null
    }
    setState('idle')
    readyAtRef.current = null
  }, [])

  const start = useCallback(async () => {
    setError(null)
    setState('connecting')
    try {
      const transport = new SmallWebRTCTransport({
        iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],
        offerUrlTemplate: SERVER_OFFER_URL,
      })
      const options: PipecatClientOptions = {
        transport,
        enableMic: true,
        callbacks: {
          'bot-transport-ready': () => {
            readyAtRef.current = performance.now()
          },
          'bot-ready': () => {
            setState('connected')
            readyAtRef.current = performance.now()
          },
          'user-connected': () => {
            readyAtRef.current = performance.now()
          },
          // Latency: capture the metrics frame the server emits (TASK-02-06).
          metric: (m: { name?: string; value?: number }) => {
            if (m?.name === 'e2e_latency_ms' && typeof m.value === 'number') {
              setLatency({ e2eMs: m.value, label: 'ASR→TTS first audio' })
            }
          },
          // Transcript (optional display).
          'bot-transcription': (data: { text?: string }) => {
            const text = data?.text
            if (text) {
              setTranscripts((prev) => [
                ...prev,
                { role: 'assistant', text, ts: Date.now() },
              ])
            }
          },
          'user-transcription': (data: { text?: string }) => {
            const text = data?.text
            if (text) {
              setTranscripts((prev) => [
                ...prev,
                { role: 'user', text, ts: Date.now() },
              ])
            }
          },
        } as any,
      }

      const client = new PipecatClient(options)
      clientRef.current = client
      // initDevices triggers mic permission; connect() opens the WebRTC session.
      await client.initDevices()
      await client.connect()
    } catch (e: any) {
      setError(e?.message ?? String(e))
      setState('error')
    }
  }, [])

  useEffect(() => {
    return () => {
      void stop()
    }
  }, [stop])

  return { state, error, transcripts, latency, start, stop }
}