// THIS MODULE IS TRANSPORT-AGNOSTIC AND IS THE ONLY IMPLEMENTATION OF THE TURN LOOP.
// Both avatar.js (inline gr.HTML) and avatar-iframe.js (postMessage) import it.
// Do NOT add turn behaviour to either transport file - add it here, or the iframe
// fallback silently loses the feature. tests/test_transport_seam.py enforces this.
//
// The renderer is reachable only through the seven-method stagePort. The host bridge is
// reachable only through getServer(), which is a GETTER rather than a value so this
// module never holds a host object and can be constructed before the bridge exists.
//
// Push-to-talk landed in wave 4. Note that mic.js and asr.js are imported HERE and
// constructed lazily by this module rather than being handed in by a transport: mic
// capture and ASR run in the parent document under BOTH transports, and the iframe
// transport has no AudioContext of its own to lend. Constructing them here means the
// two transport files needed zero lines of change to gain push-to-talk, which is the
// strongest possible form of the guarantee the seam exists to give. Both are still
// injectable through the factory so a harness can substitute a different model.
//
// The turn itself landed in wave 5, in this file and nowhere else, so the same holds:
// dispatchTurn, replay and requestSlower work under both transports because there is
// exactly one implementation of each. The bridge object getServer() returns exposes
// the host-registered functions (turn, greeting) as async methods; this module calls
// them with ONE argument each, because the host's bridge packs multiple arguments into
// a list and the far side would receive that list as a single positional.
//
// Plan 01-11: every entry point below that a tap can reach - dispatchTurn, replay,
// requestSlower, startListening - calls stagePort.unlockAudio() as its FIRST statement,
// before any await. The facade wraps these methods synchronously, so that statement runs
// inside the gesture's own call stack; anything after an await does not. Browsers that
// gate audio behind a gesture (iOS Safari; Chromium in a cross-origin embed) refuse a
// resume() issued after the server round trip, which is exactly where the only other
// resume() lives (audio-queue.js), and the owner's phone was silent for that reason.

import { createAsr } from './asr.js';
import { createMic, isHallucination, REJECT } from './mic.js';

/** VOICEVOX speedScale for the "Slower" re-read. Divides every phoneme length. */
export const SLOWER_SPEED = 0.75;

/**
 * @param {object}   opts
 * @param {object}   opts.stagePort   the renderer boundary
 * @param {Function} [opts.emit]      the facade's event bus
 * @param {Function} [opts.getServer] returns the host bridge, or null when there is none
 * @param {object}   [opts.mic]       overrides the mic built from avatar/mic.js
 * @param {object}   [opts.asr]       overrides the ASR built from avatar/asr.js
 * @param {object}   [opts.asrOptions] model/dtype/device overrides for the default ASR
 * @param {object}   [opts.micOptions] capture overrides for the default mic
 */
export function createTurnLoop({
  stagePort,
  emit = () => {},
  getServer = () => null,
  mic = null,
  asr = null,
  asrOptions = {},
  micOptions = {},
} = {}) {
  if (!stagePort) throw new Error('createTurnLoop: a stagePort is required');

  // The turn loop's own slice of __debug. The facade merges this object; it does not
  // own it, and this module does not own the facade's. Every key is initialised here
  // rather than on first use, because the parity suite compares __debug KEY SETS across
  // transports and a lazily-added key would make that comparison time-dependent.
  const state = {
    thinking: false,
    listening: false,
    speaking: false,
    lastTurnId: null,
    replayCount: 0,
    turnCount: 0,
    // The number that matters: dispatch (Enter, click or mic release) to the first
    // scheduled audio sample, in milliseconds, for the most recent turn.
    lastTurnMs: null,
    lastReplayMs: null,
    lastStageTimings: null,
    lastSubtitle: null,
    lastSpeed: null,
    lastError: null,
    asrTier: null,
    asrModel: null,
    lastTranscript: null,
    micRejectedCount: 0,
    micLastRejectReason: null,
  };

  // Set when a turn or a replay has been dispatched and its speech-start has not yet
  // been observed. speech-start is the event that closes the "thinking" window and
  // stamps lastTurnMs, so it is measured where the event arrives - the same place under
  // both transports - rather than guessed at from the speak() promise.
  let pendingDispatchAt = null;
  let pendingReplayAt = null;

  const now = () => performance.now();

  function setThinking(value) {
    const on = !!value;
    state.thinking = on;
    stagePort.setThinking(on);
    return on;
  }

  function setListening(value) {
    const on = !!value;
    state.listening = on;
    stagePort.setListening(on);
    return on;
  }

  const micInstance =
    mic ||
    createMic({
      emit,
      onListening: setListening,
      // Push-to-talk exists to make acoustic feedback impossible, so the mic refuses to
      // open while the avatar is thinking or speaking. mic.js adds the 200 ms tail after
      // speech-end on top of this.
      isBusy: () => state.thinking || state.speaking,
      ...micOptions,
    });

  const asrInstance = asr || createAsr({ emit, ...asrOptions });

  /**
   * The facade calls this for every event it fans out, under both transports - the
   * iframe transport forwards the frame's events through the same bus. It is how this
   * module learns that speech started or ended without holding a reference to the
   * audio path, which belongs to the stage.
   */
  function observe(name, data) {
    if (name === 'speech-start') {
      state.speaking = true;
      if (pendingDispatchAt !== null) {
        state.lastTurnMs = Math.round(now() - pendingDispatchAt);
        pendingDispatchAt = null;
        performance.mark('turn:speech-start');
        try {
          performance.measure('turn:dispatch-to-speech', 'turn:dispatch', 'turn:speech-start');
        } catch {
          /* a mark was cleared; the number is already in lastTurnMs */
        }
        // The thinking pose clears HERE, at speech-start, not when the response arrives:
        // decode and scheduling still sit between the two, and the face must not go idle
        // while the visitor is still waiting to hear something.
        setThinking(false);
        emit('latency', {
          turnId: state.lastTurnId,
          lastTurnMs: state.lastTurnMs,
          timings: state.lastStageTimings,
          speed: state.lastSpeed,
        });
      }
      if (pendingReplayAt !== null) {
        state.lastReplayMs = Math.round(now() - pendingReplayAt);
        pendingReplayAt = null;
        performance.mark('replay:speech-start');
      }
    } else if (name === 'speech-end') {
      state.speaking = false;
      micInstance.noteSpeechEnd();
    } else if (name === 'asr-tier' && data) {
      state.asrTier = data.tier ?? null;
      state.asrModel = data.model ?? null;
    }
  }

  function busyReason() {
    if (state.thinking) return 'the avatar is still thinking about the last turn';
    if (state.speaking) return 'the avatar is still speaking';
    return null;
  }

  function fail(where, err) {
    const message = String(err?.message ?? err);
    state.lastError = message;
    emit('error', { message, where });
    return err instanceof Error ? err : new Error(message);
  }

  /**
   * One turn: text in, speech out. Resolves at speech-end with a summary of the turn.
   *
   * Order matters and is asserted statically: the thinking pose engages BEFORE the
   * first await, because switching the pose when the response arrives would forfeit
   * the entire latency the pose exists to cover.
   *
   * @param {string} text        what the avatar should say back
   * @param {object} [opts]
   * @param {number} [opts.speed=1.0]  VOICEVOX speedScale; SLOWER_SPEED for the re-read
   * @param {boolean} [opts.greeting]  ignore text and speak the server's fixed greeting
   */
  async function dispatchTurn(text, { speed = 1.0, greeting = false } = {}) {
    stagePort.unlockAudio(); // first, synchronously: still inside the tap that got us here
    const busy = busyReason();
    if (busy) throw fail('dispatchTurn', new Error(busy));

    const bridge = getServer();
    if (!bridge || typeof bridge.turn !== 'function') {
      throw fail(
        'dispatchTurn',
        new Error('no host bridge: the standalone stage has nothing to synthesise with')
      );
    }

    performance.mark('turn:dispatch');
    const dispatchedAt = now();
    pendingDispatchAt = dispatchedAt;
    setThinking(true);
    emit('turn-start', { text: greeting ? null : text, speed, greeting });

    try {
      const directive = greeting
        ? await bridge.greeting()
        : await bridge.turn({ text: String(text ?? ''), speed });
      performance.mark('turn:response');
      const responseMs = Math.round(now() - dispatchedAt);

      // The host's client swallows an HTTP error into `undefined`, and the far side
      // answers a bad request with {error} rather than raising, so both are checked.
      if (directive === undefined || directive === null) {
        throw new Error('the host returned nothing for this turn - see its log');
      }
      if (directive.error) throw new Error(directive.error);
      if (!directive.audio_url || !Array.isArray(directive.timeline)) {
        throw new Error('the host returned a directive with no audio or no timeline');
      }

      state.turnCount += 1;
      state.lastTurnId = directive.turn_id ?? null;
      state.lastSubtitle = directive.subtitle ?? null;
      state.lastSpeed = directive.speed ?? speed;
      state.lastStageTimings = directive.timings ?? null;
      state.lastError = null;
      emit('turn', {
        turnId: state.lastTurnId,
        subtitle: state.lastSubtitle,
        speed: state.lastSpeed,
        timings: state.lastStageTimings,
        responseMs,
        greeting,
      });

      // Resolves at speech-end. speech-start arrives through observe() on the way.
      const played = await stagePort.speak({
        audioUrl: directive.audio_url,
        timeline: directive.timeline,
        subtitle: directive.subtitle,
        expression: directive.expression,
        turnId: directive.turn_id,
      });

      return {
        turnId: state.lastTurnId,
        subtitle: state.lastSubtitle,
        speed: state.lastSpeed,
        timings: state.lastStageTimings,
        responseMs,
        lastTurnMs: state.lastTurnMs,
        duration: played?.duration ?? null,
      };
    } catch (err) {
      pendingDispatchAt = null;
      setThinking(false);
      throw fail('dispatchTurn', err);
    }
  }

  return {
    state,
    getServer,
    observe,
    mic: micInstance,
    asr: asrInstance,
    setThinking,
    setListening,
    dispatchTurn,

    /**
     * Re-play the cached directive. There is deliberately no network access of any
     * kind in here: the deployed suite asserts zero requests with a browser request
     * listener, and a cache miss must fail rather than quietly re-download. The stage
     * keeps the decoded AudioBuffer and the timeline it last spoke.
     */
    async replay() {
      stagePort.unlockAudio();
      const busy = busyReason();
      if (busy) throw fail('replay', new Error(busy));
      if (state.turnCount === 0) throw fail('replay', new Error('nothing has been said yet'));
      state.replayCount += 1;
      performance.mark('replay:dispatch');
      pendingReplayAt = now();
      emit('replay', { turnId: state.lastTurnId, subtitle: state.lastSubtitle });
      try {
        return await stagePort.replayCached();
      } catch (err) {
        pendingReplayAt = null;
        throw fail('replay', err);
      }
    },

    /**
     * Re-synthesise the last utterance at SLOWER_SPEED. This IS a host round trip and
     * must be: the timeline has to be rebuilt from the re-synthesised query, because
     * speedScale divides every phoneme and a timeline scaled here would drift by exactly
     * the speed ratio against the new audio.
     */
    async requestSlower() {
      stagePort.unlockAudio(); // dispatchTurn unlocks too; this covers the early throw below
      if (!state.lastSubtitle) {
        throw fail('requestSlower', new Error('nothing to slow down yet - say something first'));
      }
      return dispatchTurn(state.lastSubtitle, { speed: SLOWER_SPEED });
    },

    /**
     * pointerdown on the push-to-talk control. The host binds the control; the behaviour
     * is here so both transports get it from one implementation.
     *
     * @returns {Promise<boolean>} whether capture actually started
     */
    async startListening() {
      // The stage's context, not the mic's: the reply to what is about to be said
      // arrives seconds after this pointerdown, and the pointerdown is the only gesture.
      stagePort.unlockAudio();
      const started = await micInstance.start();
      if (!started) {
        state.micRejectedCount = micInstance.__debug.rejectedCount;
        state.micLastRejectReason = micInstance.__debug.lastRejectReason;
      }
      return started;
    },

    /**
     * pointerup. Gate -> ASR -> 'transcript'.
     *
     * A gated-out push emits NOTHING - not an empty transcript, which every downstream
     * consumer would then have to special-case - and returns null.
     *
     * @returns {Promise<string|null>} the transcript, or null when nothing survived
     */
    async stopListening() {
      const utterance = await micInstance.stop();
      state.micRejectedCount = micInstance.__debug.rejectedCount;
      state.micLastRejectReason = micInstance.__debug.lastRejectReason;
      if (!utterance.ok) return null;

      let result;
      try {
        result = await asrInstance.transcribe(utterance.samples, utterance.sampleRate);
      } catch (err) {
        fail('stopListening', err);
        return null;
      }

      state.asrTier = asrInstance.getTier();
      state.asrModel = asrInstance.getModel();

      // Second line of defence: the audio passed the gate but Whisper still produced
      // subtitle boilerplate. Only short pushes are eligible - see mic.js.
      if (!result.text || isHallucination(result.text, utterance.durationMs)) {
        micInstance.noteTranscriptRejected(
          result.text ? REJECT.HALLUCINATION : REJECT.NO_AUDIO
        );
        state.micRejectedCount = micInstance.__debug.rejectedCount;
        state.micLastRejectReason = micInstance.__debug.lastRejectReason;
        return null;
      }

      state.lastTranscript = result.text;
      emit('transcript', {
        text: result.text,
        durationMs: Math.round(utterance.durationMs),
        tier: result.tier,
        gated: false,
      });
      return result.text;
    },
  };
}