@dtelecom/agents-js 0.1.0

package/dist/index.mjs.map

@@ -0,0 +1 @@
+{"version":3,"sources":["../src/core/voice-agent.ts","../src/room/room-connection.ts","../src/room/audio-input.ts","../src/room/audio-output.ts","../src/core/pipeline.ts","../src/core/context-manager.ts","../src/core/sentence-splitter.ts","../src/core/turn-detector.ts","../src/core/barge-in.ts"],"sourcesContent":["/**\n * VoiceAgent — top-level orchestrator for AI voice agents in dTelecom rooms.\n *\n * Wires together:\n * - RoomConnection (join room, publish audio track)\n * - Pipeline (STT -> LLM -> TTS)\n * - AudioInput (per-participant audio streams)\n * - AudioOutput (agent's published audio)\n * - RoomMemory (optional persistent memory)\n */\n\nimport { EventEmitter } from 'events';\nimport type { RemoteAudioTrack, RemoteTrackPublication, RemoteParticipant } from '@dtelecom/server-sdk-node';\nimport { RoomConnection } from '../room/room-connection';\nimport { AudioInput } from '../room/audio-input';\nimport { AudioOutput } from '../room/audio-output';\nimport { Pipeline } from './pipeline';\nimport type { AgentConfig, AgentStartOptions, DataMessageHandler } from './types';\nimport { createLogger } from '../utils/logger';\n\nconst log = createLogger('VoiceAgent');\n\nexport class VoiceAgent extends EventEmitter {\n  private readonly config: AgentConfig;\n  private connection: RoomConnection | null = null;\n  private pipeline: Pipeline | null = null;\n  private audioInputs = new Map<string, AudioInput>();\n  private audioOutput: AudioOutput | null = null;\n  private memory: import('../memory/room-memory').RoomMemory | null = null;\n  private _running = false;\n\n  constructor(config: AgentConfig) {\n    super();\n    this.config = config;\n  }\n\n  get running(): boolean {\n    return this._running;\n  }\n\n  get room() { return this.connection?.room ?? null; }\n\n  /** Enable saving raw TTS audio as WAV files to `dir` for debugging. */\n  enableAudioDump(dir: string): void {\n    this._dumpDir = dir;\n    if (this.audioOutput) {\n      this.audioOutput.dumpDir = dir;\n    }\n  }\n  private _dumpDir: string | null = null;\n\n  /**\n   * Speak text directly via TTS, bypassing the LLM.\n   * Use for greetings or announcements. Supports barge-in.\n   */\n  async say(text: string): Promise<void> {\n    if (!this.pipeline) {\n      throw new Error('Agent not started — call start() first');\n    }\n    await this.pipeline.say(text);\n  }\n\n  /** Start the agent — connect to room and begin listening. */\n  async start(options: AgentStartOptions): Promise<void> {\n    if (this._running) {\n      throw new Error('Agent is already running');\n    }\n\n    log.info(`Starting agent for room \"${options.room}\"...`);\n\n    // 1. Initialize memory (if enabled)\n    if (this.config.memory?.enabled) {\n      const { RoomMemory } = await import('../memory/room-memory');\n      this.memory = new RoomMemory({\n        dbPath: this.config.memory.dbPath ?? './data/memory.db',\n        room: options.room,\n      });\n      await this.memory.init();\n      this.memory.startSession();\n      log.info('Memory initialized');\n    }\n\n    // 2. Connect to room\n    this.connection = new RoomConnection();\n    await this.connection.connect({\n      room: options.room,\n      apiKey: options.apiKey,\n      apiSecret: options.apiSecret,\n      identity: options.identity ?? 'agent',\n      name: options.name ?? options.identity ?? 'AI Agent',\n    });\n\n    // 3. Publish audio track + start sending silence to keep it active\n    const source = await this.connection.publishAudioTrack();\n    this.audioOutput = new AudioOutput(source);\n    if (this._dumpDir) this.audioOutput.dumpDir = this._dumpDir;\n    this.audioOutput.startSilence();\n\n    // 4. Create pipeline\n    this.pipeline = new Pipeline({\n      stt: this.config.stt,\n      llm: this.config.llm,\n      tts: this.config.tts,\n      instructions: this.config.instructions,\n      audioOutput: this.audioOutput,\n      respondMode: this.config.respondMode,\n      agentName: this.config.agentName,\n      nameVariants: this.config.nameVariants,\n      memory: this.memory ?? undefined,\n    });\n\n    // Forward pipeline events\n    this.pipeline.on('transcription', (result) => this.emit('transcription', result));\n    this.pipeline.on('sentence', (text) => this.emit('sentence', text));\n    this.pipeline.on('response', (text) => this.emit('response', text));\n    this.pipeline.on('agentState', (state) => this.emit('agentState', state));\n    this.pipeline.on('error', (error) => this.emit('error', error));\n\n    // 5. Subscribe to existing remote participants\n    for (const participant of this.connection.room.remoteParticipants.values()) {\n      for (const [, pub] of participant.trackPublications) {\n        if (pub.track) {\n          this.handleTrackSubscribed(pub.track as RemoteAudioTrack, pub as RemoteTrackPublication, participant);\n        }\n      }\n    }\n\n    // 6. Listen for new tracks\n    this.connection.room.on('trackSubscribed', (track, pub, participant) => {\n      this.handleTrackSubscribed(track, pub, participant);\n    });\n\n    this.connection.room.on('trackUnsubscribed', (track, _pub, participant) => {\n      this.handleTrackUnsubscribed(track, participant);\n    });\n\n    this.connection.room.on('participantDisconnected', (participant) => {\n      this.handleParticipantDisconnected(participant);\n    });\n\n    this.connection.room.on('disconnected', (reason) => {\n      log.info(`Room disconnected: ${reason}`);\n      this.emit('disconnected', reason);\n    });\n\n    // 7. Data channel support\n    if (this.config.onDataMessage) {\n      this.setupDataChannel(this.config.onDataMessage);\n    }\n\n    this._running = true;\n    this.emit('connected');\n    log.info('Agent started and listening');\n  }\n\n  /** Stop the agent — disconnect and clean up. */\n  async stop(): Promise<void> {\n    if (!this._running) return;\n\n    log.info('Stopping agent...');\n    this._running = false;\n\n    if (this.pipeline) {\n      await this.pipeline.stop();\n      this.pipeline = null;\n    }\n\n    // End memory session (generates summary)\n    if (this.memory) {\n      try {\n        await this.memory.endSession(this.config.llm);\n        await this.memory.close();\n      } catch (err) {\n        log.error('Error closing memory:', err);\n      }\n      this.memory = null;\n    }\n\n    for (const [, input] of this.audioInputs) {\n      input.close();\n    }\n    this.audioInputs.clear();\n\n    if (this.audioOutput) {\n      this.audioOutput.stop();\n      this.audioOutput = null;\n    }\n\n    if (this.connection) {\n      await this.connection.disconnect();\n      this.connection = null;\n    }\n\n    this.emit('disconnected', 'agent_stopped');\n    log.info('Agent stopped');\n  }\n\n  private setupDataChannel(handler: DataMessageHandler): void {\n    if (!this.connection) return;\n\n    this.connection.room.on('dataReceived', (payload: Uint8Array, participant?: RemoteParticipant, _kind?: unknown, topic?: string) => {\n      const identity = participant?.identity ?? 'unknown';\n      handler(payload, identity, topic);\n    });\n\n    log.info('Data channel handler registered');\n  }\n\n  private handleTrackSubscribed(\n    track: RemoteAudioTrack,\n    _publication: RemoteTrackPublication,\n    participant: RemoteParticipant,\n  ): void {\n    const identity = participant.identity;\n    log.info(`Track subscribed from \"${identity}\" (sid=${track.sid})`);\n\n    // Track participant in memory\n    this.memory?.addParticipant(identity);\n\n    // Close existing AudioInput if this is a re-subscription\n    const existing = this.audioInputs.get(identity);\n    if (existing) {\n      log.info(`Closing old AudioInput for \"${identity}\" (re-subscription)`);\n      existing.close();\n    }\n\n    const audioInput = new AudioInput(track, identity);\n    this.audioInputs.set(identity, audioInput);\n\n    const sttStream = this.pipeline!.addParticipant(identity);\n\n    this.pipeAudioToSTT(audioInput, sttStream, identity);\n  }\n\n  private handleTrackUnsubscribed(\n    _track: RemoteAudioTrack,\n    participant: RemoteParticipant,\n  ): void {\n    const identity = participant.identity;\n    log.info(`Track unsubscribed from \"${identity}\"`);\n\n    const input = this.audioInputs.get(identity);\n    if (input) {\n      input.close();\n      this.audioInputs.delete(identity);\n    }\n  }\n\n  private handleParticipantDisconnected(participant: RemoteParticipant): void {\n    const identity = participant.identity;\n    log.info(`Participant disconnected: \"${identity}\"`);\n\n    const input = this.audioInputs.get(identity);\n    if (input) {\n      input.close();\n      this.audioInputs.delete(identity);\n    }\n\n    this.pipeline?.removeParticipant(identity);\n  }\n\n  private async pipeAudioToSTT(\n    input: AudioInput,\n    sttStream: { sendAudio(pcm16: Buffer): void },\n    identity: string,\n  ): Promise<void> {\n    try {\n      for await (const buffer of input.frames()) {\n        if (!this._running) break;\n        sttStream.sendAudio(buffer);\n      }\n    } catch (err) {\n      if (this._running) {\n        log.error(`Audio pipe error for \"${identity}\":`, err);\n      }\n    }\n  }\n}\n","import { Room, LocalAudioTrack, AudioSource, TrackSource } from '@dtelecom/server-sdk-node';\nimport { createLogger } from '../utils/logger';\n\nconst log = createLogger('RoomConnection');\n\nexport interface RoomConnectionOptions {\n  room: string;\n  apiKey: string;\n  apiSecret: string;\n  identity?: string;\n  name?: string;\n}\n\nexport class RoomConnection {\n  readonly room: Room;\n  private audioSource: AudioSource | null = null;\n  private localTrack: LocalAudioTrack | null = null;\n  private _connected = false;\n\n  constructor() {\n    this.room = new Room();\n  }\n\n  get connected(): boolean {\n    return this._connected;\n  }\n\n  /**\n   * Connect to a dTelecom room.\n   *\n   * 1. Create an Ed25519 JWT via AccessToken\n   * 2. Discover nearest SFU via getWsUrl()\n   * 3. Connect Room via WebRTC\n   * 4. Publish an audio track for the agent to speak through\n   */\n  async connect(options: RoomConnectionOptions): Promise<void> {\n    const { room: roomName, apiKey, apiSecret, identity = 'agent', name } = options;\n\n    log.info(`Connecting to room \"${roomName}\" as \"${identity}\"...`);\n\n    // Dynamic import to avoid bundling server-sdk-js in the main chunk\n    const { AccessToken } = await import('@dtelecom/server-sdk-js');\n\n    // Create token\n    const token = new AccessToken(apiKey, apiSecret, {\n      identity,\n      name: name ?? identity,\n    });\n    token.addGrant({\n      roomJoin: true,\n      room: roomName,\n      canPublish: true,\n      canSubscribe: true,\n      canPublishData: true,\n    });\n\n    // Discover SFU\n    const wsUrl = await token.getWsUrl();\n    const jwt = token.toJwt();\n\n    log.info(`SFU URL: ${wsUrl}`);\n\n    // Connect\n    await this.room.connect(wsUrl, jwt, { autoSubscribe: true });\n    this._connected = true;\n\n    log.info('Connected successfully');\n  }\n\n  /**\n   * Publish an audio track so the agent can speak.\n   * Returns the AudioSource to feed PCM16 audio into.\n   */\n  async publishAudioTrack(): Promise<AudioSource> {\n    if (this.audioSource) return this.audioSource;\n\n    // 48kHz mono — matches Opus/WebRTC native rate, no resampling needed\n    this.audioSource = new AudioSource(48000, 1);\n    this.localTrack = LocalAudioTrack.createAudioTrack('agent-voice', this.audioSource);\n\n    await this.room.localParticipant.publishTrack(this.localTrack, {\n      name: 'agent-voice',\n      source: TrackSource.MICROPHONE,\n    });\n\n    log.info('Audio track published');\n    return this.audioSource;\n  }\n\n  /** Disconnect from the room and clean up resources. */\n  async disconnect(): Promise<void> {\n    if (!this._connected) return;\n\n    if (this.localTrack) {\n      await this.room.localParticipant.unpublishTrack(this.localTrack);\n      this.localTrack = null;\n    }\n\n    if (this.audioSource) {\n      this.audioSource.destroy();\n      this.audioSource = null;\n    }\n\n    await this.room.disconnect();\n    this._connected = false;\n\n    log.info('Disconnected from room');\n  }\n}\n","import { RemoteAudioTrack, AudioStream, AudioFrame } from '@dtelecom/server-sdk-node';\nimport { createLogger } from '../utils/logger';\n\nconst log = createLogger('AudioInput');\n\nexport class AudioInput {\n  readonly participantIdentity: string;\n  private stream: AudioStream;\n  private _closed = false;\n  private frameCount = 0;\n\n  constructor(track: RemoteAudioTrack, participantIdentity: string) {\n    this.participantIdentity = participantIdentity;\n    // 16kHz mono — standard for STT\n    this.stream = track.createStream(16000, 1);\n    log.info(`AudioInput created for \"${participantIdentity}\" (trackSid=${track.sid})`);\n  }\n\n  get closed(): boolean {\n    return this._closed;\n  }\n\n  /**\n   * Async iterate over PCM16 buffers from this participant.\n   * Each yielded Buffer is 16kHz mono PCM16 LE.\n   */\n  async *frames(): AsyncGenerator<Buffer> {\n    for await (const frame of this.stream) {\n      if (this._closed) break;\n      this.frameCount++;\n      if (this.frameCount === 1 || this.frameCount % 500 === 0) {\n        log.info(`[${this.participantIdentity}] frame #${this.frameCount}`);\n      }\n      yield frame.toBuffer();\n    }\n    log.info(`[${this.participantIdentity}] frame iterator ended (total: ${this.frameCount})`);\n  }\n\n  /** Async iterate over AudioFrame objects. */\n  async *audioFrames(): AsyncGenerator<AudioFrame> {\n    for await (const frame of this.stream) {\n      if (this._closed) break;\n      yield frame;\n    }\n  }\n\n  close(): void {\n    if (this._closed) return;\n    this._closed = true;\n    this.stream.close();\n    log.debug(`AudioInput closed for participant \"${this.participantIdentity}\"`);\n  }\n}\n","import { AudioSource, AudioFrame } from '@dtelecom/server-sdk-node';\nimport { writeFileSync, appendFileSync, existsSync, mkdirSync } from 'fs';\nimport { join } from 'path';\nimport { createLogger } from '../utils/logger';\n\nconst log = createLogger('AudioOutput');\n\n/** Rate at which we write audio (16kHz mono, 20ms frames = 320 samples) */\nconst SAMPLE_RATE = 16000;\nconst CHANNELS = 1;\nconst FRAME_DURATION_MS = 20;\nconst SAMPLES_PER_FRAME = (SAMPLE_RATE * FRAME_DURATION_MS) / 1000; // 320 at 16kHz\n\n/** Pre-allocated silence frame */\nconst SILENCE = new Int16Array(SAMPLES_PER_FRAME);\n\nexport class AudioOutput {\n  private source: AudioSource;\n  private _playing = false;\n  private _responding = false;\n  private _stopped = false;\n  private silenceInterval: ReturnType<typeof setInterval> | null = null;\n\n  /** When set, raw PCM from TTS is saved to this directory as WAV files for debugging. */\n  dumpDir: string | null = null;\n  private dumpCounter = 0;\n\n  constructor(source: AudioSource) {\n    this.source = source;\n  }\n\n  get playing(): boolean {\n    return this._playing;\n  }\n\n  /**\n   * Mark the start of a multi-sentence response.\n   * Suppresses silence injection between sentences so partial frames\n   * in AudioSource's buffer don't get corrupted by interleaved silence.\n   */\n  beginResponse(): void {\n    this._responding = true;\n  }\n\n  /** Mark the end of a response — re-enable silence keepalive. */\n  endResponse(): void {\n    this._responding = false;\n  }\n\n  /**\n   * Start sparse silence keepalive to prevent the SFU from dropping the track.\n   * With Opus DTX enabled, the encoder handles silence natively — we only need\n   * an occasional packet to keep the SSRC alive.\n   */\n  startSilence(): void {\n    if (this.silenceInterval) return;\n\n    log.debug('Starting silence keepalive (sparse, 3s interval)');\n\n    this.silenceInterval = setInterval(() => {\n      if (!this._playing && !this._responding && !this._stopped) {\n        const f = new AudioFrame(SILENCE, SAMPLE_RATE, CHANNELS, SAMPLES_PER_FRAME);\n        this.source.captureFrame(f).catch(() => {});\n      }\n    }, 3000);\n  }\n\n  /**\n   * Write a PCM16 buffer to the audio output.\n   * The buffer is split into 20ms frames and fed to AudioSource.\n   */\n  async writeBuffer(pcm16: Buffer): Promise<void> {\n    this._playing = true;\n    try {\n      await this.writeFrames(pcm16);\n    } finally {\n      this._playing = false;\n    }\n  }\n\n  /**\n   * Write a stream of PCM16 buffers (from TTS) to the audio output.\n   * Supports cancellation via AbortSignal.\n   */\n  async writeStream(\n    stream: AsyncIterable<Buffer>,\n    signal?: AbortSignal,\n  ): Promise<void> {\n    this._playing = true;\n    const streamStart = performance.now();\n    let chunkCount = 0;\n    let totalBytes = 0;\n\n    log.debug('writeStream: started');\n\n    // Collect raw TTS chunks for WAV dump if enabled\n    const rawChunks: Buffer[] | null = this.dumpDir ? [] : null;\n\n    try {\n      for await (const chunk of stream) {\n        if (signal?.aborted) {\n          log.debug(`writeStream: cancelled after ${chunkCount} chunks, ${(performance.now() - streamStart).toFixed(0)}ms`);\n          break;\n        }\n        chunkCount++;\n        totalBytes += chunk.byteLength;\n        rawChunks?.push(Buffer.from(chunk));\n        await this.writeFrames(chunk);\n      }\n    } finally {\n      this._playing = false;\n      const elapsed = performance.now() - streamStart;\n      const audioDurationMs = (totalBytes / 2) / SAMPLE_RATE * 1000;\n      log.info(\n        `writeStream: done — ${chunkCount} chunks, ${totalBytes} bytes, ` +\n        `audio=${audioDurationMs.toFixed(0)}ms, wall=${elapsed.toFixed(0)}ms`,\n      );\n\n      // Save raw TTS audio as WAV for debugging\n      if (rawChunks && rawChunks.length > 0 && this.dumpDir) {\n        try {\n          if (!existsSync(this.dumpDir)) mkdirSync(this.dumpDir, { recursive: true });\n          const filePath = join(this.dumpDir, `tts-raw-${++this.dumpCounter}.wav`);\n          writeWav(filePath, rawChunks, SAMPLE_RATE);\n          log.info(`writeStream: saved raw TTS to ${filePath}`);\n        } catch (err) {\n          log.warn('writeStream: failed to save WAV dump:', err);\n        }\n      }\n    }\n  }\n\n  /**\n   * Split a PCM16 buffer into 20ms frames and write them at real-time pace.\n   * Partial frames at the end are sent directly — AudioSource handles\n   * accumulation in its internal buffer.\n   */\n  private async writeFrames(pcm16: Buffer): Promise<void> {\n    // Ensure aligned buffer for Int16Array.\n    // ws library may deliver Buffers with odd byteOffset.\n    const aligned = Buffer.alloc(pcm16.byteLength);\n    pcm16.copy(aligned);\n    const samples = new Int16Array(\n      aligned.buffer,\n      aligned.byteOffset,\n      aligned.byteLength / 2,\n    );\n\n    let offset = 0;\n    while (offset < samples.length) {\n      const end = Math.min(offset + SAMPLES_PER_FRAME, samples.length);\n      const frameSamples = samples.subarray(offset, end);\n\n      const frame = new AudioFrame(\n        frameSamples,\n        SAMPLE_RATE,\n        CHANNELS,\n        frameSamples.length,\n      );\n\n      await this.source.captureFrame(frame);\n\n      // Only pace full frames — partial frames don't produce an Opus packet,\n      // they just accumulate in AudioSource's buffer. Sleeping for them\n      // causes audio to play slower than real-time.\n      if (frameSamples.length === SAMPLES_PER_FRAME) {\n        await sleep(FRAME_DURATION_MS);\n      }\n\n      offset = end;\n    }\n  }\n\n  /**\n   * Write silence frames for the given duration.\n   * Used to pad the end of a response so the last Opus frame is fully flushed\n   * and the audio doesn't cut off abruptly.\n   */\n  async writeSilence(durationMs: number): Promise<void> {\n    const frameCount = Math.ceil(durationMs / FRAME_DURATION_MS);\n    for (let i = 0; i < frameCount; i++) {\n      const frame = new AudioFrame(SILENCE, SAMPLE_RATE, CHANNELS, SAMPLES_PER_FRAME);\n      await this.source.captureFrame(frame);\n      await sleep(FRAME_DURATION_MS);\n    }\n  }\n\n  /** Flush any buffered audio in AudioSource */\n  flush(): void {\n    this.source.flush();\n    this._playing = false;\n  }\n\n  /** Stop the silence keepalive */\n  stop(): void {\n    this._stopped = true;\n    if (this.silenceInterval) {\n      clearInterval(this.silenceInterval);\n      this.silenceInterval = null;\n    }\n  }\n}\n\nfunction sleep(ms: number): Promise<void> {\n  return new Promise((resolve) => setTimeout(resolve, ms));\n}\n\n/** Write a WAV file header + PCM data for debugging. */\nfunction writeWav(filePath: string, pcmChunks: Buffer[], sampleRate: number): void {\n  const dataSize = pcmChunks.reduce((sum, b) => sum + b.byteLength, 0);\n  const header = Buffer.alloc(44);\n  header.write('RIFF', 0);\n  header.writeUInt32LE(36 + dataSize, 4);\n  header.write('WAVE', 8);\n  header.write('fmt ', 12);\n  header.writeUInt32LE(16, 16);       // fmt chunk size\n  header.writeUInt16LE(1, 20);        // PCM format\n  header.writeUInt16LE(1, 22);        // mono\n  header.writeUInt32LE(sampleRate, 24);\n  header.writeUInt32LE(sampleRate * 2, 28); // byte rate\n  header.writeUInt16LE(2, 32);        // block align\n  header.writeUInt16LE(16, 34);       // bits per sample\n  header.write('data', 36);\n  header.writeUInt32LE(dataSize, 40);\n\n  writeFileSync(filePath, header);\n  for (const chunk of pcmChunks) {\n    appendFileSync(filePath, chunk);\n  }\n}\n","/**\n * Pipeline — coordinates the STT -> LLM -> TTS flow.\n *\n * Uses a producer/consumer pattern:\n * - Producer: LLM tokens -> sentence splitter -> sentence queue\n * - Consumer: sentence queue -> TTS -> audio output\n * Both run concurrently so audio playback never blocks LLM consumption.\n *\n * Supports barge-in (interruption cancels both producer and consumer).\n */\n\nimport { EventEmitter } from 'events';\nimport type {\n  STTPlugin,\n  STTStream,\n  LLMPlugin,\n  TTSPlugin,\n  TranscriptionResult,\n  RespondMode,\n  PipelineOptions,\n  PipelineEvents,\n  AgentState,\n} from './types';\nimport { ContextManager } from './context-manager';\nimport { SentenceSplitter } from './sentence-splitter';\nimport { TurnDetector } from './turn-detector';\nimport { BargeIn } from './barge-in';\nimport type { AudioOutput } from '../room/audio-output';\nimport type { RoomMemory } from '../memory/room-memory';\nimport { createLogger } from '../utils/logger';\n\nconst log = createLogger('Pipeline');\n\n/**\n * Estimated latency from AudioSource.captureFrame() to the client hearing it:\n * Opus encode → RTP → SFU → client → jitter buffer → decode.\n * We delay the \"speaking: false\" emission by this amount so the UI status\n * matches what the user actually hears.\n */\nconst AUDIO_DRAIN_MS = 800;\n\nfunction sleep(ms: number): Promise<void> {\n  return new Promise((resolve) => setTimeout(resolve, ms));\n}\n\nexport class Pipeline extends EventEmitter {\n  private readonly stt: STTPlugin;\n  private readonly llm: LLMPlugin;\n  private readonly tts: TTSPlugin | undefined;\n  private readonly audioOutput: AudioOutput;\n  private readonly context: ContextManager;\n  private readonly turnDetector: TurnDetector;\n  private readonly bargeIn: BargeIn;\n  private readonly splitter: SentenceSplitter;\n  private readonly respondMode: RespondMode;\n  private readonly agentName: string;\n  private readonly nameVariants: string[];\n  private readonly beforeRespond?: (speaker: string, text: string) => boolean | Promise<boolean>;\n  private readonly memory?: RoomMemory;\n\n  /** Active STT streams, keyed by participant identity */\n  private sttStreams = new Map<string, STTStream>();\n\n  private _processing = false;\n  private _running = false;\n  private _agentState: AgentState = 'idle';\n  /** Queued turn while current one is still processing */\n  private pendingTurn: { speaker: string; text: string } | null = null;\n\n  constructor(options: PipelineOptions) {\n    super();\n    this.stt = options.stt;\n    this.llm = options.llm;\n    this.tts = options.tts;\n    this.audioOutput = options.audioOutput;\n    this.respondMode = options.respondMode ?? 'always';\n    this.agentName = (options.agentName ?? 'assistant').toLowerCase();\n    this.nameVariants = (options.nameVariants ?? []).map((n) => n.toLowerCase());\n    this.beforeRespond = options.beforeRespond;\n    this.memory = options.memory;\n    this.context = new ContextManager({\n      instructions: options.instructions,\n    });\n    this.turnDetector = new TurnDetector({\n      silenceTimeoutMs: options.silenceTimeoutMs,\n    });\n    this.bargeIn = new BargeIn();\n    this.splitter = new SentenceSplitter();\n\n    this.turnDetector.onTurnEnd = () => {};\n\n    this.bargeIn.onInterrupt = () => {\n      this.audioOutput.flush();\n      this.splitter.reset();\n      this.setAgentState('idle');\n    };\n\n    // Warm up LLM\n    if (this.llm.warmup) {\n      this.llm.warmup(options.instructions).catch((err: unknown) => {\n        log.warn('LLM warmup failed:', err);\n      });\n    }\n\n    // Warm up TTS\n    if (this.tts?.warmup) {\n      this.tts.warmup().catch((err: unknown) => {\n        log.warn('TTS warmup failed:', err);\n      });\n    }\n  }\n\n  get processing(): boolean {\n    return this._processing;\n  }\n\n  get running(): boolean {\n    return this._running;\n  }\n\n  get agentState(): AgentState {\n    return this._agentState;\n  }\n\n  private setAgentState(state: AgentState): void {\n    if (this._agentState !== state) {\n      this._agentState = state;\n      this.emit('agentState', state);\n    }\n  }\n\n  addParticipant(identity: string): STTStream {\n    const existing = this.sttStreams.get(identity);\n    if (existing) {\n      existing.close();\n      this.sttStreams.delete(identity);\n      log.info(`Replacing STT stream for \"${identity}\"`);\n    }\n\n    const stream = this.stt.createStream();\n    this.sttStreams.set(identity, stream);\n    this._running = true;\n\n    stream.on('transcription', (result) => {\n      this.handleTranscription(identity, result);\n    });\n\n    stream.on('error', (error) => {\n      log.error(`STT error for ${identity}:`, error);\n      this.emit('error', error);\n    });\n\n    log.info(`STT stream started for participant \"${identity}\"`);\n    return stream;\n  }\n\n  async removeParticipant(identity: string): Promise<void> {\n    const stream = this.sttStreams.get(identity);\n    if (stream) {\n      await stream.close();\n      this.sttStreams.delete(identity);\n      log.info(`STT stream removed for participant \"${identity}\"`);\n    }\n  }\n\n  async stop(): Promise<void> {\n    this._running = false;\n    this.turnDetector.reset();\n    this.bargeIn.reset();\n    this.splitter.reset();\n\n    for (const [, stream] of this.sttStreams) {\n      await stream.close();\n    }\n    this.sttStreams.clear();\n\n    log.info('Pipeline stopped');\n  }\n\n  getContextManager(): ContextManager {\n    return this.context;\n  }\n\n  private lastFinalAt = 0;\n  private lastSttDuration = 0;\n\n  private async handleTranscription(speaker: string, result: TranscriptionResult): Promise<void> {\n    this.emit('transcription', { ...result, speaker });\n\n    // Non-empty interim → user is speaking\n    if (!result.isFinal && result.text.trim()) {\n      this.setAgentState('listening');\n    }\n\n    if (this.audioOutput.playing && result.text.trim().length > 0) {\n      this.bargeIn.trigger();\n    }\n\n    if (result.isFinal && result.text.trim()) {\n      const text = result.text.trim();\n      this.lastFinalAt = performance.now();\n      this.lastSttDuration = result.sttDuration ?? 0;\n\n      // Store every turn to memory (async, non-blocking)\n      this.memory?.storeTurn(speaker, text, false);\n\n      if (await this.shouldRespond(speaker, text)) {\n        this.processTurn(speaker, text);\n      } else {\n        log.info(`Not responding to \"${speaker}\": \"${text.slice(0, 60)}\" (mode=${this.respondMode})`);\n        this.setAgentState('idle');\n      }\n    } else if (result.isFinal) {\n      // Empty final or no text — user stopped speaking without a usable turn\n      this.setAgentState('idle');\n    }\n  }\n\n  /**\n   * Determine if the agent should respond to this turn.\n   * In 'always' mode: responds to everything.\n   * In 'addressed' mode: only when agent name is mentioned + optional beforeRespond hook.\n   */\n  private async shouldRespond(speaker: string, text: string): Promise<boolean> {\n    if (this.respondMode === 'always') return true;\n\n    // Check if agent name or variants are mentioned\n    const lower = text.toLowerCase();\n    const nameMatch = lower.includes(this.agentName) ||\n      this.nameVariants.some((v) => lower.includes(v));\n\n    if (!nameMatch) return false;\n\n    // If beforeRespond hook exists, let it decide\n    if (this.beforeRespond) {\n      return this.beforeRespond(speaker, text);\n    }\n\n    return true;\n  }\n\n  private async processTurn(speaker: string, text: string): Promise<void> {\n    if (this._processing) {\n      log.info(`Queuing turn (current still processing): \"${text}\"`);\n      this.pendingTurn = { speaker, text };\n      this.bargeIn.trigger();\n      return;\n    }\n\n    this._processing = true;\n\n    // ── Latency tracking ──\n    const tSpeechEnd = this.lastFinalAt;\n    const sttDuration = this.lastSttDuration;\n    let tLlmFirstToken = 0;\n    let tFirstSentence = 0;\n    let tFirstAudioPlayed = 0;\n\n    log.info(`Processing turn from \"${speaker}\": ${text}`);\n\n    try {\n      this.context.addUserTurn(speaker, text);\n\n      if (this.context.shouldSummarize()) {\n        await this.context.summarize(this.llm);\n      }\n\n      const signal = this.bargeIn.startCycle();\n\n      // Search memory for relevant past context\n      let memoryContext = '';\n      if (this.memory) {\n        try {\n          memoryContext = await this.memory.searchRelevant(text);\n        } catch (err) {\n          log.warn('Memory search failed:', err);\n        }\n      }\n\n      const messages = this.context.buildMessages(memoryContext || undefined);\n      let fullResponse = '';\n\n      this.setAgentState('thinking');\n\n      // ── Producer/Consumer pattern ──\n      const sentenceQueue: string[] = [];\n      let producerDone = false;\n      let wakeConsumer: (() => void) | null = null;\n\n      const wake = () => { wakeConsumer?.(); };\n\n      // ── Producer: consume LLM stream, split into sentences ──\n      const producer = async () => {\n        let isFirstToken = true;\n        let isFirstSentence = true;\n\n        const llmStream = this.llm.chat(messages, signal);\n        try {\n          while (!signal.aborted) {\n            const { value: chunk, done } = await llmStream.next();\n            if (done || !chunk) break;\n            if (signal.aborted) break;\n\n            if (chunk.type === 'token' && chunk.token) {\n              if (isFirstToken) {\n                tLlmFirstToken = performance.now();\n                isFirstToken = false;\n                log.info(`llm_first_token: ${(tLlmFirstToken - tSpeechEnd).toFixed(0)}ms`);\n              }\n\n              fullResponse += chunk.token;\n\n              const sentences = this.splitter.push(chunk.token);\n              for (const sentence of sentences) {\n                if (signal.aborted) break;\n                if (isFirstSentence) {\n                  tFirstSentence = performance.now();\n                  isFirstSentence = false;\n                  log.info(`first_sentence: ${(tFirstSentence - tSpeechEnd).toFixed(0)}ms — \"${sentence.slice(0, 60)}\"`);\n                }\n                sentenceQueue.push(sentence);\n                wake();\n              }\n            }\n          }\n        } finally {\n          await llmStream.return(undefined);\n        }\n\n        // Flush remaining text from splitter\n        if (!signal.aborted) {\n          const remaining = this.splitter.flush();\n          if (remaining) {\n            if (isFirstSentence) {\n              tFirstSentence = performance.now();\n              isFirstSentence = false;\n              log.info(`first_sentence (flush): ${(tFirstSentence - tSpeechEnd).toFixed(0)}ms — \"${remaining.slice(0, 60)}\"`);\n            }\n            sentenceQueue.push(remaining);\n            wake();\n          }\n        }\n\n        producerDone = true;\n        wake();\n      };\n\n      // ── Consumer: synthesize sentences and play audio ──\n      // beginResponse/endResponse suppresses silence injection between\n      // sentences so partial frames in AudioSource don't get corrupted.\n      const consumer = async () => {\n        this.audioOutput.beginResponse();\n        try {\n          while (true) {\n            if (signal.aborted) break;\n\n            if (sentenceQueue.length > 0) {\n              const sentence = sentenceQueue.shift()!;\n              // Skip sentences with no word characters (e.g. stray quotes/punctuation)\n              if (!/\\w/.test(sentence)) {\n                log.debug(`Skipping non-word sentence: \"${sentence}\"`);\n                continue;\n              }\n              await this.synthesizeAndPlay(sentence, signal, (t) => {\n                if (!tFirstAudioPlayed) {\n                  tFirstAudioPlayed = t;\n                  this.setAgentState('speaking');\n                }\n                this.emit('sentence', sentence);\n              });\n              continue;\n            }\n\n            if (producerDone) break;\n\n            // Wait for producer to push a sentence\n            await new Promise<void>((resolve) => {\n              wakeConsumer = resolve;\n            });\n            wakeConsumer = null;\n          }\n        } finally {\n          if (!signal.aborted) {\n            await this.audioOutput.writeSilence(40);\n          }\n          this.audioOutput.endResponse();\n        }\n      };\n\n      await Promise.all([producer(), consumer()]);\n\n      // ── Latency summary ──\n      // STT:  last interim (≈ end of speech) → final transcript received\n      // LLM:  final transcript → first complete sentence (TTFT + accumulation)\n      // TTS:  first sentence ready → first audio chunk to WebRTC\n      // Overall: STT + LLM + TTS\n      const ttftMs = tLlmFirstToken ? tLlmFirstToken - tSpeechEnd : 0;\n      const llmMs = tFirstSentence ? tFirstSentence - tSpeechEnd : 0;\n      const ttsMs = tFirstAudioPlayed && tFirstSentence ? tFirstAudioPlayed - tFirstSentence : 0;\n      const overallMs = sttDuration + llmMs + ttsMs;\n\n      log.info(\n        `LATENCY \"${text.slice(0, 30)}\": ` +\n        `STT=${sttDuration.toFixed(0)}ms ` +\n        `LLM=${llmMs.toFixed(0)}ms (TTFT=${ttftMs.toFixed(0)}ms) ` +\n        `TTS=${ttsMs.toFixed(0)}ms ` +\n        `Overall=${overallMs.toFixed(0)}ms`,\n      );\n\n      if (fullResponse.trim()) {\n        this.context.addAgentTurn(fullResponse.trim());\n        this.memory?.storeTurn('assistant', fullResponse.trim(), true);\n        this.emit('response', fullResponse.trim());\n      }\n\n      // Wait for audio pipeline to drain before signaling \"listening\"\n      // (AudioSource → Opus → RTP → SFU → client decode)\n      await sleep(AUDIO_DRAIN_MS);\n      this.setAgentState('idle');\n    } catch (err: unknown) {\n      if (err instanceof Error && err.name === 'AbortError') {\n        log.debug('Turn processing aborted (barge-in)');\n      } else {\n        log.error('Error processing turn:', err);\n        this.emit('error', err instanceof Error ? err : new Error(String(err)));\n      }\n    } finally {\n      this._processing = false;\n      this.bargeIn.reset();\n\n      if (this.pendingTurn) {\n        const { speaker: nextSpeaker, text: nextText } = this.pendingTurn;\n        this.pendingTurn = null;\n        log.info(`Processing queued turn from \"${nextSpeaker}\": ${nextText}`);\n        this.processTurn(nextSpeaker, nextText);\n      }\n    }\n  }\n\n  /**\n   * Speak text directly via TTS, bypassing the LLM.\n   * Supports barge-in — if the student speaks, the greeting is cut short.\n   * Adds the text to conversation context so the LLM knows what was said.\n   */\n  async say(text: string): Promise<void> {\n    if (this._processing) {\n      log.warn('say() called while processing — skipping');\n      return;\n    }\n\n    this._processing = true;\n    log.info(`say(): \"${text.slice(0, 60)}\"`);\n\n    try {\n      const signal = this.bargeIn.startCycle();\n      this.audioOutput.beginResponse();\n      this.setAgentState('thinking');\n\n      await this.synthesizeAndPlay(text, signal, () => {\n        this.setAgentState('speaking');\n        this.emit('sentence', text);\n      });\n\n      if (!signal.aborted) {\n        await this.audioOutput.writeSilence(40);\n        this.context.addAgentTurn(text);\n        this.memory?.storeTurn('assistant', text, true);\n        this.emit('response', text);\n      }\n\n      // Wait for audio pipeline to drain before signaling \"listening\"\n      await sleep(AUDIO_DRAIN_MS);\n      this.setAgentState('idle');\n    } catch (err: unknown) {\n      if (err instanceof Error && err.name === 'AbortError') {\n        log.debug('say() aborted (barge-in)');\n      } else {\n        log.error('Error in say():', err);\n        this.emit('error', err instanceof Error ? err : new Error(String(err)));\n      }\n    } finally {\n      this._processing = false;\n      this.audioOutput.endResponse();\n      this.bargeIn.reset();\n\n      if (this.pendingTurn) {\n        const { speaker: nextSpeaker, text: nextText } = this.pendingTurn;\n        this.pendingTurn = null;\n        log.info(`Processing queued turn from \"${nextSpeaker}\": ${nextText}`);\n        this.processTurn(nextSpeaker, nextText);\n      }\n    }\n  }\n\n  private async synthesizeAndPlay(\n    text: string,\n    signal: AbortSignal,\n    onFirstAudio: (timestamp: number) => void,\n  ): Promise<void> {\n    if (!this.tts || signal.aborted) {\n      log.info(`[Agent says]: ${text}`);\n      return;\n    }\n\n    try {\n      const ttsStart = performance.now();\n      let firstChunk = true;\n      let ttsChunkCount = 0;\n\n      const ttsStream = this.tts.synthesize(text, signal);\n      const measuredStream = async function* () {\n        for await (const chunk of ttsStream) {\n          ttsChunkCount++;\n          if (firstChunk) {\n            firstChunk = false;\n            const now = performance.now();\n            log.info(`tts_first_audio: ${(now - ttsStart).toFixed(0)}ms for \"${text.slice(0, 40)}\"`);\n            onFirstAudio(now);\n          }\n          yield chunk;\n        }\n      };\n\n      await this.audioOutput.writeStream(measuredStream(), signal);\n      log.info(`synthesizeAndPlay done: ${(performance.now() - ttsStart).toFixed(0)}ms, ${ttsChunkCount} chunks for \"${text.slice(0, 40)}\"`);\n    } catch (err: unknown) {\n      if (err instanceof Error && err.name === 'AbortError') return;\n      throw err;\n    }\n  }\n}\n","import type { Message, LLMPlugin } from './types';\nimport { createLogger } from '../utils/logger';\n\nconst log = createLogger('ContextManager');\n\n/** Rough token estimate: 1 token ~ 4 chars */\nfunction estimateTokens(text: string): number {\n  return Math.ceil(text.length / 4);\n}\n\nexport interface ContextManagerOptions {\n  /** System instructions for the agent */\n  instructions: string;\n  /** Max tokens before triggering summarization (default: 5000) */\n  maxContextTokens?: number;\n  /** Number of recent turns to keep verbatim (default: 4) */\n  recentTurnsToKeep?: number;\n}\n\ninterface Turn {\n  speaker: string;\n  text: string;\n  isAgent: boolean;\n  timestamp: number;\n}\n\nexport class ContextManager {\n  private readonly instructions: string;\n  private readonly maxContextTokens: number;\n  private readonly recentTurnsToKeep: number;\n\n  private turns: Turn[] = [];\n  private summary: string | null = null;\n\n  constructor(options: ContextManagerOptions) {\n    this.instructions = options.instructions;\n    this.maxContextTokens = options.maxContextTokens ?? 5000;\n    this.recentTurnsToKeep = options.recentTurnsToKeep ?? 4;\n  }\n\n  /** Add a user's speech turn to the conversation */\n  addUserTurn(speaker: string, text: string): void {\n    this.turns.push({\n      speaker,\n      text,\n      isAgent: false,\n      timestamp: Date.now(),\n    });\n  }\n\n  /** Add the agent's response to the conversation */\n  addAgentTurn(text: string): void {\n    this.turns.push({\n      speaker: 'assistant',\n      text,\n      isAgent: true,\n      timestamp: Date.now(),\n    });\n  }\n\n  /**\n   * Build the messages array for the LLM call.\n   *\n   * Structure:\n   * [system prompt]\n   * [memory context, if provided]\n   * [conversation summary, if any]\n   * [recent verbatim turns]\n   *\n   * @param memoryContext - Optional relevant context injected by the application\n   */\n  buildMessages(memoryContext?: string): Message[] {\n    const messages: Message[] = [];\n\n    // System prompt\n    messages.push({ role: 'system', content: this.instructions });\n\n    // Application-injected memory context (if available)\n    if (memoryContext) {\n      messages.push({\n        role: 'system',\n        content: `Relevant context from past conversations:\\n${memoryContext}`,\n      });\n    }\n\n    // Conversation summary (if summarization has occurred)\n    if (this.summary) {\n      messages.push({\n        role: 'system',\n        content: `Conversation summary so far:\\n${this.summary}`,\n      });\n    }\n\n    // Format turns as messages\n    const turnsToInclude = this.summary\n      ? this.turns.slice(-this.recentTurnsToKeep)\n      : this.turns;\n\n    for (const turn of turnsToInclude) {\n      if (turn.isAgent) {\n        messages.push({ role: 'assistant', content: turn.text });\n      } else {\n        messages.push({\n          role: 'user',\n          content: `[${turn.speaker}]: ${turn.text}`,\n        });\n      }\n    }\n\n    return messages;\n  }\n\n  /** Check if summarization should be triggered */\n  shouldSummarize(): boolean {\n    const totalTokens = this.turns.reduce(\n      (acc, t) => acc + estimateTokens(t.text) + 10,\n      estimateTokens(this.instructions),\n    );\n    return totalTokens > this.maxContextTokens;\n  }\n\n  /**\n   * Summarize older turns using the LLM.\n   * Keeps the most recent turns verbatim.\n   */\n  async summarize(llm: LLMPlugin): Promise<void> {\n    if (this.turns.length <= this.recentTurnsToKeep) return;\n\n    const olderTurns = this.turns.slice(0, -this.recentTurnsToKeep);\n    const transcript = olderTurns\n      .map((t) => `[${t.speaker}]: ${t.text}`)\n      .join('\\n');\n\n    const summaryPrompt: Message[] = [\n      {\n        role: 'system',\n        content: 'Summarize this conversation concisely, preserving key facts, decisions, and action items.',\n      },\n      { role: 'user', content: transcript },\n    ];\n\n    let summaryText = '';\n    for await (const chunk of llm.chat(summaryPrompt)) {\n      if (chunk.type === 'token' && chunk.token) {\n        summaryText += chunk.token;\n      }\n    }\n\n    this.summary = this.summary\n      ? `${this.summary}\\n\\n${summaryText}`\n      : summaryText;\n\n    this.turns = this.turns.slice(-this.recentTurnsToKeep);\n\n    log.info(`Summarized ${olderTurns.length} turns, ${this.turns.length} recent turns kept`);\n  }\n\n  /** Get the full transcript */\n  getFullTranscript(): string {\n    return this.turns.map((t) => `[${t.speaker}]: ${t.text}`).join('\\n');\n  }\n\n  /** Reset the context */\n  reset(): void {\n    this.turns = [];\n    this.summary = null;\n  }\n}\n","/**\n * SentenceSplitter — buffers streaming LLM tokens into speakable chunks\n * for TTS synthesis.\n *\n * Split strategy:\n * 1. Sentence boundary (.!?) — always split\n * 2. Clause boundary (,;:—) — split if buffer >= MIN_CHUNK chars\n * 3. Word boundary — forced split if buffer >= MAX_CHUNK chars\n */\n\nconst MIN_CHUNK = 20;\nconst MAX_CHUNK = 150;\n\nexport class SentenceSplitter {\n  private buffer = '';\n\n  /** Add a token and get back any speakable chunks */\n  push(token: string): string[] {\n    this.buffer += token;\n    return this.extractChunks();\n  }\n\n  /** Flush any remaining text as a final chunk */\n  flush(): string | null {\n    const text = this.buffer.trim();\n    this.buffer = '';\n    return text.length > 0 ? text : null;\n  }\n\n  /** Reset the splitter */\n  reset(): void {\n    this.buffer = '';\n  }\n\n  private extractChunks(): string[] {\n    const chunks: string[] = [];\n\n    while (true) {\n      // 1. Sentence boundary (.!?) — split on complete sentences\n      const sentenceMatch = this.buffer.match(/[^.!?]*[.!?]\\s*/);\n      if (sentenceMatch && sentenceMatch.index !== undefined) {\n        const end = sentenceMatch.index + sentenceMatch[0].length;\n        const chunk = this.buffer.slice(0, end).trim();\n        if (chunk.length >= MIN_CHUNK) {\n          chunks.push(chunk);\n          this.buffer = this.buffer.slice(end);\n          continue;\n        }\n      }\n\n      // 2. Clause boundary (,;:—) if buffer is getting long\n      if (this.buffer.length >= MAX_CHUNK) {\n        const clauseMatch = this.buffer.match(/[,;:\\u2014]\\s*/);\n        if (clauseMatch && clauseMatch.index !== undefined && clauseMatch.index >= MIN_CHUNK) {\n          const end = clauseMatch.index + clauseMatch[0].length;\n          const chunk = this.buffer.slice(0, end).trim();\n          chunks.push(chunk);\n          this.buffer = this.buffer.slice(end);\n          continue;\n        }\n\n        // 3. Word boundary — forced split\n        const spaceIdx = this.buffer.lastIndexOf(' ', MAX_CHUNK);\n        if (spaceIdx >= MIN_CHUNK) {\n          const chunk = this.buffer.slice(0, spaceIdx).trim();\n          chunks.push(chunk);\n          this.buffer = this.buffer.slice(spaceIdx);\n          continue;\n        }\n      }\n\n      break;\n    }\n\n    return chunks;\n  }\n}\n","import { createLogger } from '../utils/logger';\n\nconst log = createLogger('TurnDetector');\n\nexport interface TurnDetectorOptions {\n  /** Silence duration after final transcription before triggering (default: 800ms) */\n  silenceTimeoutMs?: number;\n}\n\nexport class TurnDetector {\n  private readonly silenceTimeoutMs: number;\n  private silenceTimer: ReturnType<typeof setTimeout> | null = null;\n  private _onTurnEnd: (() => void) | null = null;\n  private lastFinalText = '';\n\n  constructor(options: TurnDetectorOptions = {}) {\n    this.silenceTimeoutMs = options.silenceTimeoutMs ?? 800;\n  }\n\n  /** Set the callback for when a turn ends */\n  set onTurnEnd(cb: (() => void) | null) {\n    this._onTurnEnd = cb;\n  }\n\n  /**\n   * Feed a transcription result.\n   * Returns true if this result represents a completed turn.\n   */\n  handleTranscription(text: string, isFinal: boolean): boolean {\n    this.clearTimer();\n\n    if (isFinal && text.trim().length > 0) {\n      this.lastFinalText = text;\n\n      // Start silence timer — if no new speech, the turn is done\n      this.silenceTimer = setTimeout(() => {\n        log.debug(`Turn ended after ${this.silenceTimeoutMs}ms silence`);\n        this._onTurnEnd?.();\n      }, this.silenceTimeoutMs);\n\n      return false;\n    }\n\n    if (!isFinal && text.trim().length > 0) {\n      // Interim result — user is still speaking, reset timer\n      this.clearTimer();\n    }\n\n    return false;\n  }\n\n  /** Force-trigger turn end */\n  forceTurnEnd(): void {\n    this.clearTimer();\n    this._onTurnEnd?.();\n  }\n\n  /** Reset state */\n  reset(): void {\n    this.clearTimer();\n    this.lastFinalText = '';\n  }\n\n  private clearTimer(): void {\n    if (this.silenceTimer) {\n      clearTimeout(this.silenceTimer);\n      this.silenceTimer = null;\n    }\n  }\n}\n","import { createLogger } from '../utils/logger';\n\nconst log = createLogger('BargeIn');\n\nexport class BargeIn {\n  private abortController: AbortController | null = null;\n  private _interrupted = false;\n  private _onInterrupt: (() => void) | null = null;\n\n  get interrupted(): boolean {\n    return this._interrupted;\n  }\n\n  /** Set the callback for when barge-in occurs */\n  set onInterrupt(cb: (() => void) | null) {\n    this._onInterrupt = cb;\n  }\n\n  /**\n   * Create a new AbortController for the current response cycle.\n   * Call this at the start of each STT->LLM->TTS cycle.\n   */\n  startCycle(): AbortSignal {\n    this.abortController = new AbortController();\n    this._interrupted = false;\n    return this.abortController.signal;\n  }\n\n  /** Trigger barge-in. Called when STT detects speech during agent output. */\n  trigger(): void {\n    if (this._interrupted) return;\n    this._interrupted = true;\n\n    log.info('Barge-in detected — cancelling current response');\n\n    if (this.abortController) {\n      this.abortController.abort();\n      this.abortController = null;\n    }\n\n    this._onInterrupt?.();\n  }\n\n  /** Reset after the interrupted cycle is cleaned up */\n  reset(): void {\n    this._interrupted = false;\n    this.abortController = null;\n  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package/dist/providers/index.d.mts

@@ -0,0 +1,142 @@
+import { j as STTPlugin, k as STTStreamOptions, S as STTStream, L as LLMPlugin, M as Message, e as LLMChunk, l as TTSPlugin } from '../types-Cs5uUoTC.mjs';
+import '@dtelecom/server-sdk-node';
+
+/**
+ * DeepgramSTT — real-time streaming STT via Deepgram WebSocket API.
+ *
+ * Protocol:
+ * - Connect to wss://api.deepgram.com/v1/listen?... with config as query params
+ * - Auth via Authorization header: "Token <apiKey>"
+ * - Send audio as binary WebSocket frames (PCM16 16kHz mono)
+ * - Receive JSON: { type: "Results", channel: { alternatives: [{ transcript }] }, is_final, speech_final }
+ * - Send KeepAlive every 5s when no audio is being sent
+ * - Send CloseStream to gracefully shut down
+ *
+ * End-of-utterance strategy:
+ *   Buffer all is_final=true transcripts. Emit the buffered utterance as a
+ *   single final TranscriptionResult when speech_final=true OR UtteranceEnd
+ *   arrives. Interim results (is_final=false) are emitted immediately for
+ *   real-time feedback.
+ */
+
+interface DeepgramSTTOptions {
+    apiKey: string;
+    /** Deepgram model (default: 'nova-3') */
+    model?: string;
+    /** Language code (default: 'en') */
+    language?: string;
+    /** Enable interim results (default: true) */
+    interimResults?: boolean;
+    /** Enable punctuation (default: true) */
+    punctuate?: boolean;
+    /** Endpointing in ms (default: 300). Set to false to disable. */
+    endpointing?: number | false;
+    /** Keywords to boost recognition (e.g. ['dTelecom:5', 'WebRTC:3']) */
+    keywords?: string[];
+    /** Enable smart formatting (default: false) */
+    smartFormat?: boolean;
+    /** Utterance end timeout in ms (default: 1000). Requires interimResults. */
+    utteranceEndMs?: number;
+}
+declare class DeepgramSTT implements STTPlugin {
+    private readonly options;
+    constructor(options: DeepgramSTTOptions);
+    createStream(options?: STTStreamOptions): STTStream;
+}
+
+/**
+ * OpenRouterLLM — streaming LLM via OpenRouter (OpenAI-compatible API).
+ *
+ * Uses native fetch() with SSE parsing for streaming responses.
+ * No SDK dependency — just HTTP.
+ */
+
+interface OpenRouterLLMOptions {
+    apiKey: string;
+    /** Model identifier (e.g. 'openai/gpt-4o', 'anthropic/claude-sonnet-4') */
+    model: string;
+    /** Max tokens in response (default: 512) */
+    maxTokens?: number;
+    /** Sampling temperature 0-2 (default: 0.7) */
+    temperature?: number;
+    /** OpenRouter provider routing preferences */
+    providerRouting?: {
+        /** Sort providers by metric (e.g. 'latency') */
+        sort?: string;
+        /** Pin to specific providers in order */
+        order?: string[];
+        /** Allow fallback to other providers if pinned ones fail */
+        allowFallbacks?: boolean;
+    };
+}
+declare class OpenRouterLLM implements LLMPlugin {
+    private readonly apiKey;
+    private readonly model;
+    private readonly maxTokens;
+    private readonly temperature;
+    private readonly provider?;
+    constructor(options: OpenRouterLLMOptions);
+    /**
+     * Warm up the LLM by sending the system prompt and a short message.
+     * Primes the HTTP/TLS connection and model loading on the provider side.
+     */
+    warmup(systemPrompt: string): Promise<void>;
+    chat(messages: Message[], signal?: AbortSignal): AsyncGenerator<LLMChunk>;
+}
+
+/**
+ * CartesiaTTS — real-time streaming TTS via Cartesia WebSocket API.
+ *
+ * Protocol:
+ * - Connect to wss://api.cartesia.ai/tts/websocket?api_key=...&cartesia_version=...
+ * - Send JSON: { model_id, transcript, voice: { mode: "id", id }, output_format, context_id }
+ * - Receive JSON: { type: "chunk", data: "<base64 PCM>" } — audio data
+ * - Receive JSON: { type: "done", context_id } — synthesis complete
+ * - Audio is base64-encoded PCM16 LE at the requested sample rate
+ *
+ * Uses a persistent WebSocket connection to avoid per-sentence handshake overhead.
+ * Each synthesize() call uses a unique context_id for multiplexing.
+ */
+
+interface CartesiaTTSOptions {
+    apiKey: string;
+    /** Cartesia voice ID */
+    voiceId: string;
+    /** Model ID (default: 'sonic-3') */
+    modelId?: string;
+    /** Output sample rate in Hz (default: 16000) */
+    sampleRate?: number;
+    /** API version (default: '2024-06-10') */
+    apiVersion?: string;
+    /** Language code (default: 'en') */
+    language?: string;
+    /** Speech speed multiplier, 0.6-1.5 (default: 1.0). Sonic-3 only. */
+    speed?: number;
+    /** Emotion string (e.g. 'friendly', 'calm'). Sonic-3 only. */
+    emotion?: string;
+}
+declare class CartesiaTTS implements TTSPlugin {
+    private readonly apiKey;
+    private readonly voiceId;
+    private readonly modelId;
+    private readonly sampleRate;
+    private readonly apiVersion;
+    private readonly language?;
+    private readonly speed;
+    private readonly emotion;
+    private ws;
+    private _connected;
+    private connectPromise;
+    /** Active contexts keyed by context_id */
+    private contexts;
+    private contextCounter;
+    constructor(options: CartesiaTTSOptions);
+    /** Pre-connect the WebSocket so first synthesize() doesn't pay connection cost. */
+    warmup(): Promise<void>;
+    synthesize(text: string, signal?: AbortSignal): AsyncGenerator<Buffer>;
+    /** Ensure the persistent WebSocket is connected. */
+    private ensureConnection;
+    private handleMessage;
+}
+
+export { CartesiaTTS, type CartesiaTTSOptions, DeepgramSTT, type DeepgramSTTOptions, OpenRouterLLM, type OpenRouterLLMOptions };

package/dist/providers/index.d.ts

@@ -0,0 +1,142 @@
+import { j as STTPlugin, k as STTStreamOptions, S as STTStream, L as LLMPlugin, M as Message, e as LLMChunk, l as TTSPlugin } from '../types-Cs5uUoTC.js';
+import '@dtelecom/server-sdk-node';
+
+/**
+ * DeepgramSTT — real-time streaming STT via Deepgram WebSocket API.
+ *
+ * Protocol:
+ * - Connect to wss://api.deepgram.com/v1/listen?... with config as query params
+ * - Auth via Authorization header: "Token <apiKey>"
+ * - Send audio as binary WebSocket frames (PCM16 16kHz mono)
+ * - Receive JSON: { type: "Results", channel: { alternatives: [{ transcript }] }, is_final, speech_final }
+ * - Send KeepAlive every 5s when no audio is being sent
+ * - Send CloseStream to gracefully shut down
+ *
+ * End-of-utterance strategy:
+ *   Buffer all is_final=true transcripts. Emit the buffered utterance as a
+ *   single final TranscriptionResult when speech_final=true OR UtteranceEnd
+ *   arrives. Interim results (is_final=false) are emitted immediately for
+ *   real-time feedback.
+ */
+
+interface DeepgramSTTOptions {
+    apiKey: string;
+    /** Deepgram model (default: 'nova-3') */
+    model?: string;
+    /** Language code (default: 'en') */
+    language?: string;
+    /** Enable interim results (default: true) */
+    interimResults?: boolean;
+    /** Enable punctuation (default: true) */
+    punctuate?: boolean;
+    /** Endpointing in ms (default: 300). Set to false to disable. */
+    endpointing?: number | false;
+    /** Keywords to boost recognition (e.g. ['dTelecom:5', 'WebRTC:3']) */
+    keywords?: string[];
+    /** Enable smart formatting (default: false) */
+    smartFormat?: boolean;
+    /** Utterance end timeout in ms (default: 1000). Requires interimResults. */
+    utteranceEndMs?: number;
+}
+declare class DeepgramSTT implements STTPlugin {
+    private readonly options;
+    constructor(options: DeepgramSTTOptions);
+    createStream(options?: STTStreamOptions): STTStream;
+}
+
+/**
+ * OpenRouterLLM — streaming LLM via OpenRouter (OpenAI-compatible API).
+ *
+ * Uses native fetch() with SSE parsing for streaming responses.
+ * No SDK dependency — just HTTP.
+ */
+
+interface OpenRouterLLMOptions {
+    apiKey: string;
+    /** Model identifier (e.g. 'openai/gpt-4o', 'anthropic/claude-sonnet-4') */
+    model: string;
+    /** Max tokens in response (default: 512) */
+    maxTokens?: number;
+    /** Sampling temperature 0-2 (default: 0.7) */
+    temperature?: number;
+    /** OpenRouter provider routing preferences */
+    providerRouting?: {
+        /** Sort providers by metric (e.g. 'latency') */
+        sort?: string;
+        /** Pin to specific providers in order */
+        order?: string[];
+        /** Allow fallback to other providers if pinned ones fail */
+        allowFallbacks?: boolean;
+    };
+}
+declare class OpenRouterLLM implements LLMPlugin {
+    private readonly apiKey;
+    private readonly model;
+    private readonly maxTokens;
+    private readonly temperature;
+    private readonly provider?;
+    constructor(options: OpenRouterLLMOptions);
+    /**
+     * Warm up the LLM by sending the system prompt and a short message.
+     * Primes the HTTP/TLS connection and model loading on the provider side.
+     */
+    warmup(systemPrompt: string): Promise<void>;
+    chat(messages: Message[], signal?: AbortSignal): AsyncGenerator<LLMChunk>;
+}
+
+/**
+ * CartesiaTTS — real-time streaming TTS via Cartesia WebSocket API.
+ *
+ * Protocol:
+ * - Connect to wss://api.cartesia.ai/tts/websocket?api_key=...&cartesia_version=...
+ * - Send JSON: { model_id, transcript, voice: { mode: "id", id }, output_format, context_id }
+ * - Receive JSON: { type: "chunk", data: "<base64 PCM>" } — audio data
+ * - Receive JSON: { type: "done", context_id } — synthesis complete
+ * - Audio is base64-encoded PCM16 LE at the requested sample rate
+ *
+ * Uses a persistent WebSocket connection to avoid per-sentence handshake overhead.
+ * Each synthesize() call uses a unique context_id for multiplexing.
+ */
+
+interface CartesiaTTSOptions {
+    apiKey: string;
+    /** Cartesia voice ID */
+    voiceId: string;
+    /** Model ID (default: 'sonic-3') */
+    modelId?: string;
+    /** Output sample rate in Hz (default: 16000) */
+    sampleRate?: number;
+    /** API version (default: '2024-06-10') */
+    apiVersion?: string;
+    /** Language code (default: 'en') */
+    language?: string;
+    /** Speech speed multiplier, 0.6-1.5 (default: 1.0). Sonic-3 only. */
+    speed?: number;
+    /** Emotion string (e.g. 'friendly', 'calm'). Sonic-3 only. */
+    emotion?: string;
+}
+declare class CartesiaTTS implements TTSPlugin {
+    private readonly apiKey;
+    private readonly voiceId;
+    private readonly modelId;
+    private readonly sampleRate;
+    private readonly apiVersion;
+    private readonly language?;
+    private readonly speed;
+    private readonly emotion;
+    private ws;
+    private _connected;
+    private connectPromise;
+    /** Active contexts keyed by context_id */
+    private contexts;
+    private contextCounter;
+    constructor(options: CartesiaTTSOptions);
+    /** Pre-connect the WebSocket so first synthesize() doesn't pay connection cost. */
+    warmup(): Promise<void>;
+    synthesize(text: string, signal?: AbortSignal): AsyncGenerator<Buffer>;
+    /** Ensure the persistent WebSocket is connected. */
+    private ensureConnection;
+    private handleMessage;
+}
+
+export { CartesiaTTS, type CartesiaTTSOptions, DeepgramSTT, type DeepgramSTTOptions, OpenRouterLLM, type OpenRouterLLMOptions };