@xmov/avatar 2.1.0 → 2.1.1

package/dist/agent/avatar.modern.js.map

@@ -1 +0,0 @@
-{"version":3,"file":"avatar.modern.js","sources":["../../src/agent/e2e-client.ts","../../src/agent/fixed-audio-track.ts","../../src/agent/types.ts","../../src/agent/audio-debug.ts","../../src/utils/audio-session.ts","../../src/utils/media-recorder-timestamp.ts","../../src/encoding/webcodec-opus-encoder.ts","../../src/encoding/webm-muxer.ts","../../src/encoding/pcm-webm-encoder.ts","../../src/agent/microphone.ts","../../src/agent/audio-uplink.ts","../../src/agent/index.ts"],"sourcesContent":["import type { AgentClientMessage, AgentServerEvent, AgentSocketState } from \"./types\";\n\nconst MAX_BUFFERED_AMOUNT = 2 * 1024 * 1024;\n\nexport interface AgentE2EClientOptions {\n  wsUrl: string;\n  token: string;\n  audioInputEnabled?: boolean;\n  WebSocketCtor?: typeof WebSocket;\n  onStateChange?: (state: AgentSocketState) => void;\n  onEvent?: (event: AgentServerEvent) => void;\n  onError?: (error: Error) => void;\n  onClose?: (event: CloseEvent) => void;\n}\n\nexport class AgentE2EClient {\n  private ws: WebSocket | null = null;\n  private connectPromise: Promise<void> | null = null;\n  private rejectConnect: ((error: Error) => void) | null = null;\n  private audioInputEnabled: boolean;\n  private readonly WebSocketCtor: typeof WebSocket;\n  private readonly options: AgentE2EClientOptions;\n\n  constructor(options: AgentE2EClientOptions) {\n    this.options = options;\n    this.audioInputEnabled = options.audioInputEnabled ?? true;\n    const WebSocketCtor = options.WebSocketCtor || globalThis.WebSocket;\n    if (!WebSocketCtor) {\n      throw new Error(\"当前环境不支持 WebSocket\");\n    }\n    this.WebSocketCtor = WebSocketCtor;\n  }\n\n  get isOpen() {\n    return this.ws?.readyState === this.WebSocketCtor.OPEN;\n  }\n\n  connect(): Promise<void> {\n    if (this.isOpen) {\n      return Promise.resolve();\n    }\n    if (this.connectPromise) {\n      return this.connectPromise;\n    }\n\n    this.options.onStateChange?.(\"connecting\");\n\n    let connection: Promise<void>;\n    connection = new Promise<void>((resolve, reject) => {\n      this.rejectConnect = reject;\n      let settled = false;\n      const ws = new this.WebSocketCtor(this.buildWsUrl(this.options.wsUrl, this.options.token));\n      this.ws = ws;\n\n      ws.onopen = () => {\n        if (this.ws !== ws) {\n          return;\n        }\n        settled = true;\n        if (this.connectPromise === connection) {\n          this.connectPromise = null;\n        }\n        this.rejectConnect = null;\n        this.options.onStateChange?.(\"open\");\n        resolve();\n      };\n\n      ws.onmessage = (message) => {\n        if (this.ws !== ws) {\n          return;\n        }\n        const event = this.parseMessage(message.data);\n        if (event) {\n          this.options.onEvent?.(event);\n        }\n      };\n\n      ws.onerror = () => {\n        if (this.ws !== ws) {\n          return;\n        }\n        const error = new Error(\"E2E WebSocket 连接异常\");\n        if (!settled) {\n          settled = true;\n          if (this.connectPromise === connection) {\n            this.connectPromise = null;\n          }\n          this.rejectConnect = null;\n          reject(error);\n        }\n        this.options.onStateChange?.(\"error\");\n        this.options.onError?.(error);\n      };\n\n      ws.onclose = (event) => {\n        if (this.ws !== ws) {\n          return;\n        }\n        this.ws = null;\n        if (!settled) {\n          settled = true;\n          if (this.connectPromise === connection) {\n            this.connectPromise = null;\n          }\n          this.rejectConnect = null;\n          reject(new Error(`E2E WebSocket 已关闭: ${event.code} ${event.reason || \"\"}`.trim()));\n        }\n        this.options.onStateChange?.(\"closed\");\n        this.options.onClose?.(event);\n      };\n    });\n    this.connectPromise = connection;\n    return this.connectPromise;\n  }\n\n  disconnect(code = 1000, reason = \"client_close\") {\n    if (!this.ws) {\n      return;\n    }\n\n    const ws = this.ws;\n    this.ws = null;\n    this.connectPromise = null;\n    const rejectConnect = this.rejectConnect;\n    this.rejectConnect = null;\n    rejectConnect?.(new Error(\"E2E WebSocket 连接已取消\"));\n\n    if (ws.readyState === this.WebSocketCtor.OPEN || ws.readyState === this.WebSocketCtor.CONNECTING) {\n      ws.close(code, reason);\n    }\n  }\n\n  send(message: AgentClientMessage) {\n    this.assertOpen();\n    this.ws?.send(JSON.stringify(message));\n  }\n\n  trySend(message: AgentClientMessage) {\n    this.assertOpen();\n    if (this.ws && this.ws.bufferedAmount > MAX_BUFFERED_AMOUNT) {\n      return false;\n    }\n    this.ws?.send(JSON.stringify(message));\n    return true;\n  }\n\n  sendBinary(frame: ArrayBuffer | ArrayBufferView | Blob) {\n    this.assertOpen();\n    if (this.ws && this.ws.bufferedAmount > MAX_BUFFERED_AMOUNT) {\n      const error = new Error(\"E2E WebSocket 发送缓冲区超限\");\n      error.name = \"WebSocketBackpressureError\";\n      throw error;\n    }\n    this.ws?.send(frame as any);\n  }\n\n  ping() {\n    this.send({ type: \"ping\" });\n  }\n\n  setAudioInputEnabled(enabled: boolean) {\n    this.audioInputEnabled = enabled;\n  }\n\n  private assertOpen() {\n    if (!this.isOpen) {\n      throw new Error(\"E2E WebSocket 未连接\");\n    }\n  }\n\n  private parseMessage(data: unknown): AgentServerEvent | null {\n    if (typeof data !== \"string\") {\n      return null;\n    }\n\n    try {\n      const parsed = JSON.parse(data);\n      if (parsed && typeof parsed.type === \"string\") {\n        return parsed as AgentServerEvent;\n      }\n      return null;\n    } catch (error) {\n      this.options.onError?.(new Error(\"E2E WebSocket 消息不是合法 JSON\"));\n      return null;\n    }\n  }\n\n  private buildWsUrl(wsUrl: string, token: string) {\n    try {\n      const url = new URL(wsUrl);\n      if (token && !url.searchParams.has(\"token\") && !url.searchParams.has(\"e2e_token\")) {\n        url.searchParams.set(\"token\", token);\n      }\n      this.updateAudioInputQuery(url.searchParams);\n      return url.toString();\n    } catch {\n      const [baseUrl, query = \"\"] = wsUrl.split(\"?\", 2);\n      const searchParams = new URLSearchParams(query);\n      if (token && !searchParams.has(\"token\") && !searchParams.has(\"e2e_token\")) {\n        searchParams.set(\"token\", token);\n      }\n      this.updateAudioInputQuery(searchParams);\n      const nextQuery = searchParams.toString();\n      return nextQuery ? `${baseUrl}?${nextQuery}` : baseUrl;\n    }\n  }\n\n  private updateAudioInputQuery(searchParams: URLSearchParams) {\n    if (!this.audioInputEnabled) {\n      searchParams.set(\"audio_input_off\", \"1\");\n      return;\n    }\n    searchParams.delete(\"audio_input_off\");\n  }\n}\n","const TARGET_SAMPLE_RATE = 16_000;\nconst TARGET_CHANNELS = 1;\nconst OUTPUT_FRAME_SIZE = 320;\nconst MIN_INPUT_SAMPLE_RATE = 8_000;\nconst MAX_INPUT_SAMPLE_RATE = 192_000;\nconst MAX_INPUT_FRAMES_PER_READ = 96_000;\nconst MAX_RESAMPLE_BUFFER_LENGTH = MAX_INPUT_FRAMES_PER_READ * 2;\n\ninterface AudioDataLike {\n  sampleRate: number;\n  numberOfChannels: number;\n  numberOfFrames: number;\n  timestamp: number;\n  copyTo(destination: Float32Array, options?: { format?: string; planeIndex?: number }): void;\n  close(): void;\n}\n\ninterface AudioDataConstructorLike {\n  new (init: {\n    format: \"f32-planar\";\n    sampleRate: number;\n    numberOfFrames: number;\n    numberOfChannels: number;\n    timestamp: number;\n    data: Float32Array;\n  }): AudioDataLike;\n}\n\ninterface AudioTrackProcessorLike {\n  readable: ReadableStream<AudioDataLike>;\n}\n\ninterface AudioTrackProcessorConstructorLike {\n  new (init: { track: MediaStreamTrack }): AudioTrackProcessorLike;\n}\n\ninterface AudioTrackGeneratorLike extends MediaStreamTrack {\n  writable: WritableStream<AudioDataLike>;\n}\n\ninterface AudioTrackGeneratorConstructorLike {\n  new (init: { kind: \"audio\" }): AudioTrackGeneratorLike;\n}\n\ninterface AudioRuntimeGlobals {\n  AudioData?: AudioDataConstructorLike;\n  MediaStreamTrackProcessor?: AudioTrackProcessorConstructorLike;\n  MediaStreamTrackGenerator?: AudioTrackGeneratorConstructorLike;\n  webkitAudioContext?: typeof AudioContext;\n}\n\ntype AgentCodedError = Error & {\n  agentCode?: string;\n  retryable?: boolean;\n  cause?: unknown;\n};\n\nexport interface FixedAudioTrackOptions {\n  onError?: (error: Error) => void;\n}\n\n/**\n * Converts a microphone MediaStreamTrack to a verifiable 16 kHz mono track\n * before it is handed to MediaRecorder. WebM/Opus remains the container/codec\n * used on the wire; this class only owns the decoded audio track conversion.\n */\nexport class FixedAudioTrack {\n  private readonly options: FixedAudioTrackOptions;\n  private inputTrack: MediaStreamTrack | null = null;\n  private reader: ReadableStreamDefaultReader<AudioDataLike> | null = null;\n  private writer: WritableStreamDefaultWriter<AudioDataLike> | null = null;\n  private generator: AudioTrackGeneratorLike | null = null;\n  private audioContext: AudioContext | null = null;\n  private audioSource: MediaStreamAudioSourceNode | null = null;\n  private audioDestination: MediaStreamAudioDestinationNode | null = null;\n  private pumpPromise: Promise<void> | null = null;\n  private pumpFinished = false;\n  private outputClosed = false;\n  private draining = false;\n  private inputSampleRate: number | null = null;\n  private resampleBuffer: number[] = [];\n  private resamplePosition = 0;\n  private outputBuffer: number[] = [];\n  private outputTimestamp: number | null = null;\n  private startGeneration = 0;\n\n  constructor(options: FixedAudioTrackOptions = {}) {\n    this.options = options;\n  }\n\n  async start(inputTrack: MediaStreamTrack): Promise<MediaStream> {\n    const startGeneration = ++this.startGeneration;\n    this.inputTrack = inputTrack;\n    this.inputSampleRate = null;\n    this.resampleBuffer = [];\n    this.resamplePosition = 0;\n    this.outputBuffer = [];\n    this.outputTimestamp = null;\n    this.pumpFinished = false;\n    this.outputClosed = false;\n    this.draining = false;\n    const globals = globalThis as typeof globalThis & AudioRuntimeGlobals;\n    const ProcessorCtor = globals.MediaStreamTrackProcessor;\n    const GeneratorCtor = globals.MediaStreamTrackGenerator;\n    const AudioDataCtor = globals.AudioData;\n    if (!ProcessorCtor || !GeneratorCtor || !AudioDataCtor) {\n      return this.startWebAudio(inputTrack, globals, startGeneration);\n    }\n\n    const processor = new ProcessorCtor({ track: inputTrack });\n    const generator = new GeneratorCtor({ kind: \"audio\" });\n    const stream = new MediaStream([generator]);\n    const outputTrack = stream.getAudioTracks()[0];\n    const settings = outputTrack?.getSettings() as (MediaTrackSettings & {\n      sampleRate?: number;\n      channelCount?: number;\n    }) | undefined;\n    // Chrome does not expose generator settings until its first AudioData is\n    // written. The format is nevertheless fixed by every AudioData created in\n    // writeChunk; when settings are available, reject an implementation that\n    // reports a different output track.\n    if (\n      (settings?.sampleRate !== undefined && settings.sampleRate !== TARGET_SAMPLE_RATE) ||\n      (settings?.channelCount !== undefined && settings.channelCount !== TARGET_CHANNELS)\n    ) {\n      generator.stop();\n      throw this.unsupported(\n        `浏览器音频输出格式不符合要求: ${settings?.sampleRate || \"unknown\"} Hz / ${settings?.channelCount || \"unknown\"} channel`,\n      );\n    }\n\n    try {\n      this.reader = processor.readable.getReader();\n      this.writer = generator.writable.getWriter();\n    } catch (error) {\n      generator.stop();\n      if (inputTrack.readyState !== \"ended\") {\n        inputTrack.stop();\n      }\n      this.inputTrack = null;\n      throw error;\n    }\n    this.generator = generator;\n    this.pumpPromise = this.pump(AudioDataCtor);\n    return stream;\n  }\n\n  private async startWebAudio(\n    inputTrack: MediaStreamTrack,\n    globals: AudioRuntimeGlobals,\n    startGeneration: number,\n  ) {\n    const AudioContextCtor = globalThis.AudioContext || globals.webkitAudioContext;\n    if (!AudioContextCtor) {\n      throw this.unsupported(\"当前浏览器不支持 16 kHz mono 音频处理\");\n    }\n\n    let context: AudioContext | null = null;\n    let source: MediaStreamAudioSourceNode | null = null;\n    let destination: MediaStreamAudioDestinationNode | null = null;\n    try {\n      context = new AudioContextCtor({ sampleRate: TARGET_SAMPLE_RATE });\n      this.audioContext = context;\n      if (context.sampleRate !== TARGET_SAMPLE_RATE) {\n        throw this.unsupported(`浏览器无法创建 16 kHz AudioContext: ${context.sampleRate} Hz`);\n      }\n      if (context.state === \"suspended\") {\n        try {\n          await context.resume();\n        } catch (cause) {\n          throw this.contextSuspended(cause);\n        }\n      }\n      this.assertStartActive(startGeneration);\n      if (context.state !== \"running\") {\n        throw this.contextSuspended();\n      }\n\n      source = context.createMediaStreamSource(new MediaStream([inputTrack]));\n      this.audioSource = source;\n      destination = context.createMediaStreamDestination();\n      this.audioDestination = destination;\n      destination.channelCount = TARGET_CHANNELS;\n      destination.channelCountMode = \"explicit\";\n      destination.channelInterpretation = \"speakers\";\n      source.connect(destination);\n\n      const outputTrack = destination.stream.getAudioTracks()[0];\n      if (!outputTrack) {\n        throw new Error(\"Web Audio 未返回音频输出轨道\");\n      }\n      this.validateOutputTrack(outputTrack);\n      this.assertStartActive(startGeneration);\n      return destination.stream;\n    } catch (error) {\n      source?.disconnect();\n      destination?.stream.getTracks().forEach((track) => track.stop());\n      if (context && context.state !== \"closed\") {\n        await context.close().catch(() => undefined);\n      }\n      if (inputTrack.readyState !== \"ended\") {\n        inputTrack.stop();\n      }\n      this.inputTrack = null;\n      if (this.audioContext === context) {\n        this.audioContext = null;\n        this.audioSource = null;\n        this.audioDestination = null;\n      }\n      throw this.normalizeWebAudioError(error);\n    }\n  }\n\n  async stop() {\n    await this.drain();\n    if (this.audioContext) {\n      await this.closeWebAudioOutput();\n      return;\n    }\n    await this.closeOutput();\n  }\n\n  /** Stop input and flush decoded samples while keeping the output track live. */\n  async drain() {\n    this.startGeneration += 1;\n    this.draining = true;\n    this.inputTrack?.stop();\n    this.inputTrack = null;\n\n    if (this.audioContext) {\n      return;\n    }\n\n    const pump = this.pumpPromise;\n    if (pump) {\n      const settled = pump.then(\n        () => true,\n        () => true,\n      );\n      const finished = await Promise.race([settled, this.delay(1000)]);\n      if (!finished) {\n        await Promise.race([\n          this.reader?.cancel().catch(() => undefined),\n          this.delay(100),\n        ]);\n        await Promise.race([\n          this.writer?.abort(new Error(\"16 kHz mono 音频处理停止超时\")).catch(() => undefined),\n          this.delay(100),\n        ]);\n        this.stopGenerator();\n        // Do not await an implementation that is stuck in writer.write(). The\n        // rejection handler above keeps the eventual pump settlement observed.\n        void settled;\n      }\n    }\n  }\n\n  private async closeOutput() {\n    const writer = this.writer;\n    this.outputClosed = true;\n    if (writer && this.pumpFinished) {\n      await Promise.race([\n        writer.close().catch(() => undefined),\n        this.delay(100),\n      ]);\n    } else {\n      await Promise.race([\n        writer?.abort(new Error(\"16 kHz mono 音频处理停止超时\")).catch(() => undefined),\n        this.delay(100),\n      ]);\n    }\n    this.stopGenerator();\n    if (writer && this.pumpFinished) {\n      writer.releaseLock();\n    }\n    this.reader = null;\n    this.writer = null;\n    this.generator = null;\n    this.pumpPromise = null;\n  }\n\n  private async closeWebAudioOutput() {\n    const context = this.audioContext;\n    const source = this.audioSource;\n    const destination = this.audioDestination;\n    this.audioContext = null;\n    this.audioSource = null;\n    this.audioDestination = null;\n\n    source?.disconnect();\n    destination?.disconnect();\n    destination?.stream.getTracks().forEach((track) => track.stop());\n    if (context && context.state !== \"closed\") {\n      await context.close().catch(() => undefined);\n    }\n  }\n\n  private async pump(AudioDataCtor: AudioDataConstructorLike) {\n    const reader = this.reader;\n    const writer = this.writer;\n    if (!reader || !writer) {\n      return;\n    }\n\n    try {\n      let inputEnded = false;\n      while (true) {\n        const result = await reader.read();\n        if (result.done) {\n          inputEnded = true;\n          break;\n        }\n        if (!result.value) {\n          continue;\n        }\n\n        const audioData = result.value;\n        try {\n          this.validateAudioData(audioData);\n          const mono = this.toMono(audioData);\n          const chunks = this.resample(mono, audioData.sampleRate, false);\n          if (this.outputTimestamp === null) {\n            this.outputTimestamp = Number.isFinite(audioData.timestamp) ? audioData.timestamp : 0;\n          }\n          for (const chunk of chunks) {\n            await this.writeChunk(writer, AudioDataCtor, chunk);\n          }\n        } finally {\n          audioData.close();\n        }\n      }\n\n      for (const chunk of this.resample(new Float32Array(), this.inputSampleRate || TARGET_SAMPLE_RATE, true)) {\n        await this.writeChunk(writer, AudioDataCtor, chunk);\n      }\n      if (inputEnded && !this.draining && this.inputTrack?.readyState === \"ended\") {\n        const error = new Error(\"麦克风音频轨道已结束\");\n        error.name = \"AudioTransformError\";\n        this.options.onError?.(error);\n        this.stopGenerator();\n      }\n    } catch (error) {\n      const normalized = error instanceof Error ? error : new Error(\"16 kHz mono 音频处理失败\");\n      normalized.name = \"AudioTransformError\";\n      this.options.onError?.(normalized);\n    } finally {\n      reader.releaseLock();\n      this.pumpFinished = true;\n      if (this.outputClosed) {\n        writer.releaseLock();\n      }\n    }\n  }\n\n  private toMono(audioData: AudioDataLike) {\n    const channels = Math.max(1, audioData.numberOfChannels);\n    const mono = new Float32Array(audioData.numberOfFrames);\n    for (let channel = 0; channel < channels; channel += 1) {\n      const plane = new Float32Array(audioData.numberOfFrames);\n      audioData.copyTo(plane, { format: \"f32-planar\", planeIndex: channel });\n      for (let index = 0; index < plane.length; index += 1) {\n        mono[index] += plane[index] / channels;\n      }\n    }\n    return mono;\n  }\n\n  private resample(input: Float32Array, inputSampleRate: number, flush: boolean) {\n    if (this.inputSampleRate === null) {\n      this.inputSampleRate = inputSampleRate;\n    } else if (this.inputSampleRate !== inputSampleRate) {\n      throw new Error(`输入音频采样率发生变化: ${this.inputSampleRate} -> ${inputSampleRate}`);\n    }\n\n    for (const sample of input) {\n      this.resampleBuffer.push(sample);\n    }\n    if (this.resampleBuffer.length > MAX_RESAMPLE_BUFFER_LENGTH) {\n      throw new Error(\"输入音频重采样缓冲区超限\");\n    }\n    if (flush && this.resampleBuffer.length > 0) {\n      this.resampleBuffer.push(this.resampleBuffer[this.resampleBuffer.length - 1]);\n    }\n\n    const step = inputSampleRate / TARGET_SAMPLE_RATE;\n    const chunks: Float32Array[] = [];\n    while (this.resamplePosition + 1 < this.resampleBuffer.length) {\n      const before = Math.floor(this.resamplePosition);\n      const after = Math.min(before + 1, this.resampleBuffer.length - 1);\n      const weight = this.resamplePosition - before;\n      this.outputBuffer.push(\n        this.resampleBuffer[before] * (1 - weight) + this.resampleBuffer[after] * weight,\n      );\n      this.resamplePosition += step;\n\n      const consumed = Math.floor(this.resamplePosition);\n      if (consumed > 0) {\n        this.resampleBuffer.splice(0, consumed);\n        this.resamplePosition -= consumed;\n      }\n\n      if (this.outputBuffer.length >= OUTPUT_FRAME_SIZE) {\n        chunks.push(new Float32Array(this.outputBuffer.splice(0, OUTPUT_FRAME_SIZE)));\n      }\n    }\n\n    if (flush && this.outputBuffer.length > 0) {\n      chunks.push(new Float32Array(this.outputBuffer.splice(0)));\n    }\n    return chunks;\n  }\n\n  private async writeChunk(\n    writer: WritableStreamDefaultWriter<AudioDataLike>,\n    AudioDataCtor: AudioDataConstructorLike,\n    samples: Float32Array,\n  ) {\n    if (!samples.length) {\n      return;\n    }\n    const timestamp = this.outputTimestamp || 0;\n    const audioData = new AudioDataCtor({\n      format: \"f32-planar\",\n      sampleRate: TARGET_SAMPLE_RATE,\n      numberOfFrames: samples.length,\n      numberOfChannels: TARGET_CHANNELS,\n      timestamp,\n      data: samples,\n    });\n    this.outputTimestamp = timestamp + Math.round((samples.length * 1_000_000) / TARGET_SAMPLE_RATE);\n    await writer.write(audioData);\n    audioData.close();\n    const settings = this.generator?.getSettings() as\n      | (MediaTrackSettings & { sampleRate?: number; channelCount?: number })\n      | undefined;\n    if (\n      settings &&\n      ((settings.sampleRate !== undefined && settings.sampleRate !== TARGET_SAMPLE_RATE) ||\n        (settings.channelCount !== undefined && settings.channelCount !== TARGET_CHANNELS))\n    ) {\n      throw this.unsupported(\n        `浏览器音频输出格式不符合要求: ${settings.sampleRate || \"unknown\"} Hz / ${settings.channelCount || \"unknown\"} channel`,\n      );\n    }\n  }\n\n  private validateAudioData(audioData: AudioDataLike) {\n    if (\n      !Number.isFinite(audioData.sampleRate) ||\n      !Number.isInteger(audioData.sampleRate) ||\n      audioData.sampleRate < MIN_INPUT_SAMPLE_RATE ||\n      audioData.sampleRate > MAX_INPUT_SAMPLE_RATE\n    ) {\n      throw new Error(`输入音频采样率无效: ${audioData.sampleRate}`);\n    }\n    if (\n      !Number.isInteger(audioData.numberOfChannels) ||\n      audioData.numberOfChannels < 1 ||\n      audioData.numberOfChannels > 8\n    ) {\n      throw new Error(`输入音频声道数无效: ${audioData.numberOfChannels}`);\n    }\n    if (\n      !Number.isInteger(audioData.numberOfFrames) ||\n      audioData.numberOfFrames < 1 ||\n      audioData.numberOfFrames > MAX_INPUT_FRAMES_PER_READ\n    ) {\n      throw new Error(`输入音频帧数无效: ${audioData.numberOfFrames}`);\n    }\n    if (!Number.isFinite(audioData.timestamp) || Math.abs(audioData.timestamp) > 1e15) {\n      throw new Error(`输入音频时间戳无效: ${audioData.timestamp}`);\n    }\n  }\n\n  private delay(timeoutMs: number) {\n    return new Promise<boolean>((resolve) => {\n      globalThis.setTimeout(() => resolve(false), timeoutMs);\n    });\n  }\n\n  private stopGenerator() {\n    if (this.generator && this.generator.readyState !== \"ended\") {\n      this.generator.stop();\n    }\n  }\n\n  private unsupported(message: string, cause?: unknown) {\n    const error = new Error(message) as AgentCodedError;\n    error.name = \"NotSupportedError\";\n    error.agentCode = \"AUDIO_FIXED_TRACK_UNSUPPORTED\";\n    error.retryable = false;\n    error.cause = cause;\n    return error;\n  }\n\n  private contextSuspended(cause?: unknown) {\n    const error = new Error(\"AudioContext 未运行，请在用户操作后重试\") as AgentCodedError;\n    error.name = \"NotAllowedError\";\n    error.agentCode = \"AUDIO_CONTEXT_SUSPENDED\";\n    error.retryable = true;\n    error.cause = cause;\n    return error;\n  }\n\n  private assertStartActive(startGeneration: number) {\n    if (startGeneration !== this.startGeneration) {\n      const error = new Error(\"16 kHz mono 音频处理启动已取消\") as AgentCodedError;\n      error.name = \"AbortError\";\n      error.agentCode = \"AUDIO_FIXED_TRACK_CANCELLED\";\n      error.retryable = true;\n      throw error;\n    }\n  }\n\n  private normalizeWebAudioError(error: unknown) {\n    const normalized: AgentCodedError = error instanceof Error\n      ? error\n      : new Error(String(error));\n    if (normalized.agentCode) {\n      return normalized;\n    }\n    if (normalized.name === \"NotSupportedError\") {\n      return this.unsupported(`浏览器无法初始化 16 kHz mono Web Audio: ${normalized.message}`, normalized);\n    }\n    const coded = new Error(`16 kHz mono Web Audio 初始化失败: ${normalized.message}`) as AgentCodedError;\n    coded.name = normalized.name;\n    coded.agentCode = \"AUDIO_FIXED_TRACK_FAILED\";\n    coded.retryable = true;\n    coded.cause = normalized;\n    return coded;\n  }\n\n  private validateOutputTrack(outputTrack: MediaStreamTrack) {\n    const settings = outputTrack.getSettings() as MediaTrackSettings & {\n      sampleRate?: number;\n      channelCount?: number;\n    };\n    if (\n      (settings.sampleRate !== undefined && settings.sampleRate !== TARGET_SAMPLE_RATE) ||\n      (settings.channelCount !== undefined && settings.channelCount !== TARGET_CHANNELS)\n    ) {\n      throw this.unsupported(\n        `浏览器音频输出格式不符合要求: ${settings.sampleRate || \"unknown\"} Hz / ${settings.channelCount || \"unknown\"} channel`,\n      );\n    }\n  }\n}\n","import type { IAvatarOptions, IInitParams, INetworkInfo, RenderState } from \"../types\";\n\nexport type AgentState =\n  | \"idle\"\n  | \"initializing\"\n  | \"ready\"\n  | \"running\"\n  | \"reconnecting\"\n  | \"stopped\"\n  | \"destroyed\"\n  | \"failed\";\nexport type AgentSocketState = \"idle\" | \"connecting\" | \"open\" | \"closed\" | \"error\";\nexport type AgentASRState = \"idle\" | \"requesting-permission\" | \"starting\" | \"listening\" | \"stopping\" | \"failed\";\nexport type AgentConversationState =\n  | \"idle\"\n  | \"asking\"\n  | \"speaking-directly\"\n  | \"speaking\"\n  | \"completed\"\n  | \"interrupted\"\n  | \"failed\";\n\nexport type AgentErrorDomain = \"permission\" | \"network\" | \"asr\" | \"brain\" | \"ttsa\" | \"quota\" | \"sdk\";\n\nexport interface AgentAudioOptions {\n  chunkMs?: number;\n  inputEnabled?: boolean;\n  /**\n   * @deprecated Session 返回 features.speech_frontend.enable_aec 后，以服务端返回值为准。\n   */\n  echoCancellationEnabled?: boolean;\n}\n\nexport interface AgentReconnectOptions {\n  enabled?: boolean;\n  maxAttempts?: number;\n  initialDelayMs?: number;\n  maxDelayMs?: number;\n}\n\nexport interface AgentError {\n  code: string;\n  message: string;\n  domain: AgentErrorDomain;\n  retryable?: boolean;\n  cause?: unknown;\n}\n\nexport interface AgentASRResult {\n  text: string;\n  isFinal: boolean;\n  raw?: unknown;\n}\n\nexport interface AgentASRStartOptions {\n  /**\n   * Optional caller-owned audio stream used instead of requesting the physical microphone.\n   * The Agent clones one live audio track and never stops the caller-owned stream.\n   */\n  inputStream?: MediaStream;\n}\n\nexport interface AgentLLMUsage {\n  promptTokens: number;\n  completionTokens: number;\n  totalTokens: number;\n  cachedTokens: number;\n}\n\nexport interface AgentLLMResponse {\n  event: \"chunk\" | \"done\";\n  text?: string;\n  isFirst?: boolean;\n  usage?: AgentLLMUsage;\n  raw?: unknown;\n}\n\nexport interface AgentSemanticJudgeResult {\n  query: string;\n  meaningful: boolean;\n  action: string;\n  raw?: unknown;\n}\n\nexport interface AgentSpeakStateEvent {\n  state: string;\n  clientSpeakId?: string | number;\n  raw?: unknown;\n}\n\nexport interface AgentConversationEvent {\n  state: AgentConversationState;\n  text?: string;\n  raw?: unknown;\n}\n\nexport interface XingyunAvatarAgentCallbacks {\n  onAgentStateChange?: (state: AgentState) => void;\n  onSocketStateChange?: (state: AgentSocketState) => void;\n  onASRStateChange?: (state: AgentASRState) => void;\n  onASRResult?: (result: AgentASRResult) => void;\n  onLLMResponse?: (response: AgentLLMResponse) => void;\n  onSpeakStateChange?: (event: AgentSpeakStateEvent) => void;\n  onConversationChange?: (event: AgentConversationEvent) => void;\n  onSemanticJudgeResult?: (result: AgentSemanticJudgeResult) => void;\n  onRenderChange?: (state: RenderState) => void;\n  onNetworkInfo?: (info: INetworkInfo) => void;\n  onError?: (error: AgentError) => void;\n}\n\nexport type AgentServiceConfig = Record<string, unknown>;\n\nexport interface AgentBrainConfig extends AgentServiceConfig {\n  extra_body?: Record<string, unknown>;\n}\n\nexport interface AgentFeatures extends AgentServiceConfig {\n  anti_interference?: {\n    semantic_judge_enabled?: boolean;\n  };\n  speech_frontend?: {\n    enabled?: boolean;\n    enable_aec?: boolean;\n    enable_speech_separation?: boolean;\n  };\n  vad_merge_mode?: boolean;\n  volume_and_repetition_text_detection?: boolean;\n}\n\nexport interface XingyunAvatarAgentOptions extends IAvatarOptions {\n  asr_id?: number;\n  asr_config?: AgentServiceConfig;\n  tts_config?: AgentServiceConfig;\n  features?: AgentFeatures;\n  extras?: AgentServiceConfig;\n  llm_id?: number;\n  brain_config?: AgentBrainConfig;\n  session_speak_req_id?: 0 | 1;\n  e2eServer?: string;\n  authToken?: string;\n  audio?: AgentAudioOptions;\n  reconnect?: AgentReconnectOptions;\n  agentCallbacks?: XingyunAvatarAgentCallbacks;\n  webSocketCtor?: typeof WebSocket;\n}\n\nexport type XingyunAvatarAgentInitParams = Omit<IInitParams, \"onDownloadProgress\"> & {\n  onDownloadProgress?: IInitParams[\"onDownloadProgress\"];\n};\n\nexport interface AgentSessionInfo {\n  ws_url?: string;\n  e2e_token?: string;\n  monitor_ws_url?: string;\n}\n\nexport interface AgentClientMessage {\n  type: string;\n  message?: unknown;\n  [key: string]: unknown;\n}\n\nexport interface AgentServerEvent {\n  type: string;\n  domain?: string;\n  code?: string | number;\n  event?: string;\n  message?: unknown;\n  text?: string;\n  is_first?: boolean;\n  is_final?: boolean;\n  usage?: {\n    prompt_tokens?: number;\n    completion_tokens?: number;\n    total_tokens?: number;\n    cached_tokens?: number;\n  };\n  [key: string]: unknown;\n}\n\nexport interface AgentAudioMetadata {\n  format: \"webm_opus\";\n  mimeType: \"audio/webm;codecs=opus\";\n  /** Fixed output track sample rate passed to the MediaRecorder pipeline. */\n  sampleRate: number;\n  /** Fixed output track channel count passed to the MediaRecorder pipeline. */\n  channels: number;\n  chunkMs: number;\n}\n\nexport const DEFAULT_AGENT_AUDIO: AgentAudioMetadata = {\n  format: \"webm_opus\",\n  mimeType: \"audio/webm;codecs=opus\",\n  sampleRate: 16000,\n  channels: 1,\n  chunkMs: 100,\n};\n","const DEFAULT_AUDIO_CHUNK_PREVIEW_BYTES = 64;\n\ntype AudioChunkDebugMetadata = Record<string, unknown>;\n\nfunction toHex(byte: number) {\n  return byte.toString(16).padStart(2, \"0\");\n}\n\nexport async function logAudioChunkContent(\n  label: string,\n  chunk: Blob | ArrayBuffer | ArrayBufferView,\n  metadata: AudioChunkDebugMetadata = {},\n  previewBytes = DEFAULT_AUDIO_CHUNK_PREVIEW_BYTES,\n) {\n  try {\n    const buffer = chunk instanceof Blob\n      ? await chunk.arrayBuffer()\n      : ArrayBuffer.isView(chunk)\n        ? chunk.buffer.slice(chunk.byteOffset, chunk.byteOffset + chunk.byteLength)\n        : chunk;\n    const bytes = new Uint8Array(buffer);\n    const preview = Array.from(bytes.slice(0, previewBytes));\n    const type = chunk instanceof Blob ? chunk.type : metadata.type;\n\n    console.log(label, {\n      ...metadata,\n      size: bytes.byteLength,\n      type,\n      previewLength: preview.length,\n      bytes: preview,\n      hex: preview.map(toHex).join(\" \"),\n    });\n  } catch (error) {\n    console.warn(label, \"failed to read audio chunk\", error);\n  }\n}\n","export type WebAudioSessionType = \"playback\" | \"play-and-record\";\n\ntype WebAudioSession = {\n  type: string;\n};\n\ntype NavigatorWithAudioSession = Navigator & {\n  audioSession?: WebAudioSession;\n};\n\ntype WebAudioPlaybackSessionRestoredListener = () => void;\n\nconst captureStates = new WeakMap<\n  WebAudioSession,\n  { count: number; previousType: string }\n>();\nconst playbackSessionRestoredListeners = new WeakMap<\n  WebAudioSession,\n  Set<WebAudioPlaybackSessionRestoredListener>\n>();\n\n/**\n * Keep Web Audio audible through the iOS Ring/Silent switch. Do not override\n * a play-and-record session that may belong to an active microphone track.\n */\nexport function configureWebAudioPlaybackSession(\n  target: Navigator = navigator,\n): boolean {\n  const audioSession = getAudioSession(target);\n  if (!audioSession) {\n    return false;\n  }\n\n  const captureState = captureStates.get(audioSession);\n  if (captureState?.count || audioSession.type === \"play-and-record\") {\n    return setAudioSessionType(audioSession, \"play-and-record\");\n  }\n  return setAudioSessionType(audioSession, \"playback\");\n}\n\nexport function beginWebAudioCaptureSession(\n  target: Navigator = navigator,\n): boolean {\n  const audioSession = getAudioSession(target);\n  if (!audioSession) {\n    return false;\n  }\n\n  const existingState = captureStates.get(audioSession);\n  const previousType = existingState?.previousType ?? audioSession.type;\n  if (!setAudioSessionType(audioSession, \"play-and-record\")) {\n    return false;\n  }\n\n  captureStates.set(audioSession, {\n    count: (existingState?.count ?? 0) + 1,\n    previousType,\n  });\n  return true;\n}\n\nexport function subscribeWebAudioPlaybackSessionRestored(\n  listener: WebAudioPlaybackSessionRestoredListener,\n  target: Navigator = navigator,\n): () => void {\n  const audioSession = getAudioSession(target);\n  if (!audioSession) {\n    return () => {};\n  }\n\n  let listeners = playbackSessionRestoredListeners.get(audioSession);\n  if (!listeners) {\n    listeners = new Set();\n    playbackSessionRestoredListeners.set(audioSession, listeners);\n  }\n  listeners.add(listener);\n\n  return () => {\n    listeners?.delete(listener);\n    if (listeners?.size === 0) {\n      playbackSessionRestoredListeners.delete(audioSession);\n    }\n  };\n}\n\nexport function endWebAudioCaptureSession(\n  target: Navigator = navigator,\n): boolean {\n  const audioSession = getAudioSession(target);\n  if (!audioSession) {\n    return false;\n  }\n\n  const captureState = captureStates.get(audioSession);\n  if (!captureState) {\n    return configureWebAudioPlaybackSession(target);\n  }\n\n  captureState.count -= 1;\n  if (captureState.count > 0) {\n    return setAudioSessionType(audioSession, \"play-and-record\");\n  }\n\n  captureStates.delete(audioSession);\n  const nextType = captureState.previousType === \"play-and-record\"\n    ? \"play-and-record\"\n    : \"playback\";\n  const updated = setAudioSessionType(audioSession, nextType);\n  if (updated && nextType === \"playback\") {\n    notifyPlaybackSessionRestored(audioSession);\n  }\n  return updated;\n}\n\nfunction notifyPlaybackSessionRestored(audioSession: WebAudioSession): void {\n  const listeners = playbackSessionRestoredListeners.get(audioSession);\n  if (!listeners) {\n    return;\n  }\n\n  for (const listener of [...listeners]) {\n    try {\n      listener();\n    } catch {\n      // Playback recovery must never break microphone cleanup.\n    }\n  }\n}\n\nfunction getAudioSession(target: Navigator): WebAudioSession | undefined {\n  return (target as NavigatorWithAudioSession).audioSession;\n}\n\nfunction setAudioSessionType(\n  audioSession: WebAudioSession,\n  type: WebAudioSessionType,\n): boolean {\n  try {\n    audioSession.type = type;\n    return audioSession.type === type;\n  } catch {\n    return false;\n  }\n}\n","type EpochClock = () => number;\n\nfunction monotonicEpochNow() {\n  const clock = globalThis.performance;\n  if (\n    clock\n    && Number.isFinite(clock.timeOrigin)\n    && typeof clock.now === \"function\"\n  ) {\n    return clock.timeOrigin + clock.now();\n  }\n  return Date.now();\n}\n\n/** Normalizes browser-specific BlobEvent.timecode units to epoch milliseconds. */\nexport class MediaRecorderTimestamp {\n  private readonly startedAt: number;\n  private readonly now: EpochClock;\n  private timecodeScale: 1 | 1000 | null = null;\n  private firstTimecode: number | null = null;\n  private lastTimecode: number | null = null;\n  private lastObservedAt: number | null = null;\n  private lastTimestamp: number | null = null;\n\n  constructor(\n    private readonly chunkMs: number,\n    now: EpochClock = monotonicEpochNow,\n  ) {\n    this.now = now;\n    this.startedAt = now();\n  }\n\n  resolve(timecode: number): number {\n    const observedAt = this.now();\n    const validTimecode = Number.isFinite(timecode) && timecode >= 0;\n\n    if (validTimecode && this.firstTimecode === null) {\n      this.firstTimecode = timecode;\n    }\n\n    if (\n      validTimecode\n      && this.timecodeScale === null\n      && this.lastTimecode !== null\n      && timecode > this.lastTimecode\n    ) {\n      const timecodeDelta = timecode - this.lastTimecode;\n      const observedDelta = this.lastObservedAt === null ? 0 : observedAt - this.lastObservedAt;\n      const expectedDelta = observedDelta >= this.chunkMs / 4 ? observedDelta : this.chunkMs;\n      const millisecondError = Math.abs(timecodeDelta - expectedDelta);\n      const secondError = Math.abs(timecodeDelta * 1000 - expectedDelta);\n      this.timecodeScale = secondError < millisecondError ? 1000 : 1;\n    }\n\n    let timestamp = observedAt - this.chunkMs;\n    const relativeTimecode = validTimecode && this.firstTimecode !== null\n      ? timecode - this.firstTimecode\n      : null;\n    if (\n      relativeTimecode !== null\n      && relativeTimecode >= 0\n      && (this.timecodeScale !== null || relativeTimecode === 0)\n    ) {\n      timestamp = this.startedAt + relativeTimecode * (this.timecodeScale ?? 1);\n    }\n\n    if (this.lastTimestamp !== null) {\n      // A non-empty MediaRecorder Blob must not collapse into an already-used\n      // AU bucket when a browser repeats or resets its timecode.\n      timestamp = Math.max(timestamp, this.lastTimestamp + this.chunkMs);\n    }\n\n    const resolved = Math.max(Math.round(this.startedAt), Math.round(timestamp));\n    this.lastTimecode = validTimecode ? timecode : this.lastTimecode;\n    this.lastObservedAt = observedAt;\n    this.lastTimestamp = resolved;\n    return resolved;\n  }\n}\n","/**\n * WebCodecs AudioEncoder Opus 编码封装\n * 使用浏览器原生 WebCodecs API 将 PCM 音频编码为 Opus 帧\n */\n\nexport interface OpusEncoderConfig {\n  sampleRate: number;      // 16000\n  channels: number;        // 1 (mono)\n  bitrate: number;         // 24000\n  frameSize: number;       // 1600 (100ms @ 16kHz)\n}\n\nexport interface EncodedOpusFrame {\n  data: Uint8Array;\n  durationMs: number;\n}\n\nconst OPUS_FRAME_DURATION_US = 20_000;\nconst ENCODE_TIMEOUT_MS = 2_000;\n\ninterface PendingEncode {\n  startTimestampUs: number;\n  endTimestampUs: number;\n  expectedFrameCount: number;\n  chunks: EncodedAudioChunk[];\n  resolve(frames: EncodedOpusFrame[]): void;\n  reject(error: Error): void;\n  timeoutId: number;\n}\n\nfunction buildAudioEncoderConfig(config: OpusEncoderConfig) {\n  return {\n    codec: 'opus',\n    sampleRate: config.sampleRate,\n    numberOfChannels: config.channels,\n    bitrate: config.bitrate,\n    // Safari may otherwise choose a 120 ms Opus packet for a 100 ms AudioData\n    // input. The AU protocol clock is fixed at 100 ms, so pin packetization to\n    // five 20 ms packets per input block.\n    opus: {\n      format: 'opus' as const,\n      frameDuration: OPUS_FRAME_DURATION_US,\n    },\n  };\n}\n\nexport class WebCodecOpusEncoder {\n  private encoder: AudioEncoder | null = null;\n  private config: OpusEncoderConfig;\n  private isInitialized = false;\n  private pendingEncode: PendingEncode | null = null;\n  private timestamp: number = 0; // 单位：微秒\n\n  constructor(config: Partial<OpusEncoderConfig> = {}) {\n    this.config = {\n      sampleRate: 16000,\n      channels: 1,\n      bitrate: 24000,\n      frameSize: 1600,\n      ...config,\n    };\n  }\n\n  /**\n   * 检测当前浏览器是否支持 WebCodecs Opus 编码\n   */\n  static async isSupported(): Promise<boolean> {\n    if (typeof AudioEncoder === 'undefined') return false;\n    try {\n      const support = await AudioEncoder.isConfigSupported(buildAudioEncoderConfig({\n        sampleRate: 16000,\n        channels: 1,\n        bitrate: 24000,\n        frameSize: 1600,\n      }));\n      return support.supported === true\n        && support.config?.opus?.frameDuration === OPUS_FRAME_DURATION_US;\n    } catch {\n      return false;\n    }\n  }\n\n  async init(): Promise<void> {\n    if (this.isInitialized) return;\n\n    this.encoder = this.createEncoder();\n\n    this.isInitialized = true;\n  }\n\n  /**\n   * 异步编码 PCM 为 Opus 帧\n   * @param pcm S16LE PCM 数据 (Int16Array, 1600 samples = 100ms)\n   * @returns 保留 packet 边界和时长的 Opus 帧数组\n   */\n  async encode(pcm: Int16Array): Promise<EncodedOpusFrame[]> {\n    if (!this.isInitialized || !this.encoder) {\n      return Promise.reject(new Error('WebCodecOpusEncoder 未初始化'));\n    }\n    if (this.pendingEncode) {\n      return Promise.reject(new Error('WebCodecs Opus 编码器仍在处理上一音频块'));\n    }\n\n    // 构造 AudioData（S16 格式）\n    // 复制为独立 ArrayBuffer，避免 SharedArrayBuffer 类型不兼容\n    // 同时确保数据边界精确（Int16Array 可能是更大 buffer 的 view）\n    const pcmCopy = new Int16Array(pcm);\n\n    const startTimestampUs = this.timestamp;\n    const blockDurationUs = Math.round((pcm.length * 1_000_000) / this.config.sampleRate);\n    const endTimestampUs = startTimestampUs + blockDurationUs;\n    const expectedFrameCount = blockDurationUs / OPUS_FRAME_DURATION_US;\n    if (!Number.isInteger(expectedFrameCount)) {\n      return Promise.reject(new Error('PCM 音频块无法按 20ms Opus packet 完整分包'));\n    }\n\n    const audioData = new AudioData({\n      format: 's16',\n      sampleRate: this.config.sampleRate,\n      numberOfFrames: pcmCopy.length,\n      numberOfChannels: this.config.channels,\n      timestamp: startTimestampUs,\n      data: pcmCopy,\n    });\n\n    const result = new Promise<EncodedOpusFrame[]>((resolve, reject) => {\n      const timeoutId = globalThis.setTimeout(() => {\n        this.failPending(new Error('WebCodecs Opus 编码输出超过 2 秒'));\n      }, ENCODE_TIMEOUT_MS);\n      this.pendingEncode = {\n        startTimestampUs,\n        endTimestampUs,\n        expectedFrameCount,\n        chunks: [],\n        resolve,\n        reject,\n        timeoutId,\n      };\n    });\n\n    try {\n      this.encoder.encode(audioData);\n      this.timestamp = endTimestampUs;\n    } catch (cause) {\n      this.failPending(cause instanceof Error ? cause : new Error('WebCodecs Opus 编码失败'));\n    } finally {\n      audioData.close();\n    }\n\n    return result;\n  }\n\n  /**\n   * 重置时间戳（新 speak 段开始时调用）\n   */\n  reset(): void {\n    this.timestamp = 0;\n    this.failPending(new Error('WebCodecs Opus 编码已重置'));\n    if (this.encoder) {\n      try {\n        this.encoder.close();\n      } catch {}\n      this.encoder = this.createEncoder();\n    }\n  }\n\n  async destroy(): Promise<void> {\n    const encoder = this.encoder;\n    if (encoder) {\n      try {\n        await encoder.flush();\n      } catch {}\n      this.failPending(new Error('WebCodecs Opus 编码器已销毁'));\n      try {\n        encoder.close();\n      } catch {}\n    }\n    this.encoder = null;\n    this.isInitialized = false;\n  }\n\n  private createEncoder() {\n    let encoder: AudioEncoder;\n    encoder = new AudioEncoder({\n      output: (chunk) => {\n        if (this.encoder === encoder) {\n          this.handleOutput(chunk);\n        }\n      },\n      error: (cause) => {\n        if (this.encoder !== encoder) return;\n        const error = cause instanceof Error ? cause : new Error('WebCodecs Opus 编码失败');\n        const logger = (globalThis as typeof globalThis & {\n          avatarSDKLogger?: Pick<Console, 'error'>;\n        }).avatarSDKLogger || console;\n        logger.error('[WebCodecOpusEncoder] 编码错误:', error);\n        this.failPending(error);\n      },\n    });\n    encoder.configure(buildAudioEncoderConfig(this.config));\n    return encoder;\n  }\n\n  private handleOutput(chunk: EncodedAudioChunk) {\n    const pending = this.pendingEncode;\n    if (!pending) {\n      // flush() may emit one codec-delay/padding packet after all real PCM input.\n      return;\n    }\n    if (\n      chunk.timestamp < pending.startTimestampUs\n      || chunk.timestamp >= pending.endTimestampUs\n      || (chunk.duration !== null && chunk.duration !== OPUS_FRAME_DURATION_US)\n    ) {\n      this.failPending(new Error('WebCodecs Opus 未按 20ms 分包，无法生成固定 100ms AU 音频'));\n      return;\n    }\n\n    pending.chunks.push(chunk);\n    if (pending.chunks.length > pending.expectedFrameCount) {\n      this.failPending(new Error('WebCodecs Opus 未按 20ms 分包，无法生成固定 100ms AU 音频'));\n      return;\n    }\n    if (pending.chunks.length < pending.expectedFrameCount) {\n      return;\n    }\n\n    const frames = pending.chunks.map((encodedChunk, index) => {\n      const data = new Uint8Array(encodedChunk.byteLength);\n      encodedChunk.copyTo(data);\n      const nextTimestamp = pending.chunks[index + 1]?.timestamp ?? pending.endTimestampUs;\n      const durationUs = encodedChunk.duration && encodedChunk.duration > 0\n        ? encodedChunk.duration\n        : Math.max(1, nextTimestamp - encodedChunk.timestamp);\n      return { data, durationMs: durationUs / 1000 };\n    });\n    const totalDurationMs = frames.reduce((total, frame) => total + frame.durationMs, 0);\n    if (\n      frames.some((frame) => frame.durationMs !== OPUS_FRAME_DURATION_US / 1000)\n      || totalDurationMs !== (pending.endTimestampUs - pending.startTimestampUs) / 1000\n    ) {\n      this.failPending(new Error('WebCodecs Opus 未按 20ms 分包，无法生成固定 100ms AU 音频'));\n      return;\n    }\n\n    globalThis.clearTimeout(pending.timeoutId);\n    this.pendingEncode = null;\n    pending.resolve(frames);\n  }\n\n  private failPending(error: Error) {\n    const pending = this.pendingEncode;\n    if (!pending) return;\n    globalThis.clearTimeout(pending.timeoutId);\n    this.pendingEncode = null;\n    pending.reject(error);\n  }\n}\n","/**\n * WebM EBML 封装器\n * 将 Opus 帧封装为 WebM 容器格式，产出与 MSE 模式一致的 WebM/Opus 数据\n *\n * WebM 结构：\n * EBML Header → Segment → SegmentInfo → Tracks → Cluster(s) → SimpleBlock(s)\n */\n\n// EBML Element IDs\nconst EBML_ID = {\n  EBML: 0x1A45DFA3,\n  EBMLVersion: 0x4286,\n  EBMLReadVersion: 0x42F7,\n  EBMLMaxIDLength: 0x42F2,\n  EBMLMaxSizeLength: 0x42F3,\n  DocType: 0x4282,\n  DocTypeVersion: 0x4287,\n  DocTypeReadVersion: 0x4285,\n\n  Segment: 0x18538067,\n  SegmentInfo: 0x1549A966,\n  TimecodeScale: 0x2AD7B1,\n  MuxingApp: 0x4D80,\n  WritingApp: 0x5741,\n\n  Tracks: 0x1654AE6B,\n  TrackEntry: 0xAE,\n  TrackNumber: 0xD7,\n  TrackUID: 0x73C5,\n  TrackType: 0x83,\n  FlagLacing: 0x9C,\n  Language: 0x22B59C,\n  CodecID: 0x86,\n  CodecPrivate: 0x63A2,\n  CodecDelay: 0x56AA,\n  SeekPreRoll: 0x56BB,\n  Audio: 0xE1,\n  SamplingFrequency: 0xB5,\n  Channels: 0x9F,\n\n  Cluster: 0x1F43B675,\n  Timecode: 0xE7,\n  SimpleBlock: 0xA3,\n};\n\nexport class WebMMuxer {\n  private chunks: Uint8Array[] = [];\n  private hasHeader = false;\n  private clusterTimestamp: number = 0; // 单位：毫秒\n  private sampleRate: number;\n  private frameDuration: number = 100; // 100ms per frame\n\n  constructor(sampleRate: number = 16000) {\n    this.sampleRate = sampleRate;\n  }\n\n  /**\n   * 写入 WebM 文件头（EBML Header + Segment Info + Track）\n   */\n  writeHeader(): void {\n    if (this.hasHeader) return;\n\n    const header = this.buildEBMLHeader();\n    const segmentInfo = this.buildSegmentInfo();\n    const tracks = this.buildOpusTracks();\n\n    // EBML Header\n    this.chunks.push(header);\n    // Segment (open-ended, use unknown size)\n    this.chunks.push(new Uint8Array([0x18, 0x53, 0x80, 0x67, 0x01, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF, 0xFF]));\n    // SegmentInfo\n    this.chunks.push(segmentInfo);\n    // Tracks\n    this.chunks.push(tracks);\n\n    this.hasHeader = true;\n    this.clusterTimestamp = 0;\n  }\n\n  /**\n   * 写入 Opus 帧\n   * @param opusFrame Opus 编码后的数据\n   * @param durationMs 帧时长 (默认 100ms)\n   */\n  feedOpusFrame(opusFrame: Uint8Array, durationMs: number = this.frameDuration): void {\n    if (!this.hasHeader) {\n      this.writeHeader();\n    }\n\n    // 每帧都创建一个新 Cluster（与 MediaRecorder 行为一致）\n    const cluster = this.buildCluster(Math.round(this.clusterTimestamp), opusFrame);\n    this.chunks.push(cluster);\n    this.clusterTimestamp += durationMs;\n  }\n\n  /**\n   * 获取新增的 WebM 数据（自上次 flush 以来的数据）\n   * 返回后清空已返回的数据，避免重复发送\n   */\n  flush(): ArrayBuffer {\n    if (this.chunks.length === 0) {\n      return new ArrayBuffer(0);\n    }\n    \n    const totalLength = this.chunks.reduce((sum, chunk) => sum + chunk.length, 0);\n    const result = new Uint8Array(totalLength);\n    let offset = 0;\n    for (const chunk of this.chunks) {\n      result.set(chunk, offset);\n      offset += chunk.length;\n    }\n    \n    // 关键：清空已返回的数据，避免下次 flush 重复发送\n    this.chunks = [];\n    \n    return result.buffer;\n  }\n\n  /**\n   * 重置（新 speak 段开始时调用）\n   */\n  reset(): void {\n    this.chunks = [];\n    this.hasHeader = false;\n    this.clusterTimestamp = 0;\n  }\n\n  // ── EBML 编码辅助方法 ──\n\n  /**\n   * 编码 VINT（Variable-length Integer）\n   */\n  private encodeVINT(value: number, width?: number): Uint8Array {\n    if (width !== undefined) {\n      const bytes = new Uint8Array(width);\n      for (let i = width - 1; i >= 0; i--) {\n        bytes[i] = value & 0xFF;\n        value >>>= 8;\n      }\n      bytes[0] |= (1 << (8 - width));\n      return bytes;\n    }\n\n    // 自动选择最小宽度\n    if (value < 0x7F) {\n      return new Uint8Array([0x80 | value]);\n    } else if (value < 0x3FFF) {\n      return new Uint8Array([0x40 | (value >> 8), value & 0xFF]);\n    } else if (value < 0x1FFFFF) {\n      return new Uint8Array([0x20 | (value >> 16), (value >> 8) & 0xFF, value & 0xFF]);\n    } else if (value < 0x0FFFFFFF) {\n      return new Uint8Array([0x10 | (value >> 24), (value >> 16) & 0xFF, (value >> 8) & 0xFF, value & 0xFF]);\n    }\n    throw new Error(`VINT value too large: ${value}`);\n  }\n\n  /**\n   * 编码 EBML Element ID\n   */\n  private encodeID(id: number): Uint8Array {\n    if (id <= 0xFF) return new Uint8Array([id]);\n    if (id <= 0xFFFF) return new Uint8Array([id >> 8, id & 0xFF]);\n    if (id <= 0xFFFFFF) return new Uint8Array([id >> 16, (id >> 8) & 0xFF, id & 0xFF]);\n    return new Uint8Array([id >>> 24, (id >> 16) & 0xFF, (id >> 8) & 0xFF, id & 0xFF]);\n  }\n\n  /**\n   * 编码无符号整数\n   */\n  private encodeUint(value: number): Uint8Array {\n    if (value === 0) return new Uint8Array([0]);\n    const bytes: number[] = [];\n    let v = value;\n    while (v > 0) {\n      bytes.unshift(v & 0xFF);\n      v = Math.floor(v / 256);\n    }\n    return new Uint8Array(bytes);\n  }\n\n  /**\n   * 编码字符串\n   */\n  private encodeString(str: string): Uint8Array {\n    return new TextEncoder().encode(str);\n  }\n\n  /**\n   * 编码浮点数（32-bit IEEE 754）\n   */\n  private encodeFloat32(value: number): Uint8Array {\n    const buf = new ArrayBuffer(4);\n    new DataView(buf).setFloat32(0, value, false); // big-endian\n    return new Uint8Array(buf);\n  }\n\n  /**\n   * 编码 EBML Element = ID + Size + Data\n   */\n  private encodeElement(id: number, data: Uint8Array): Uint8Array {\n    const idBytes = this.encodeID(id);\n    const sizeBytes = this.encodeVINT(data.length);\n    const result = new Uint8Array(idBytes.length + sizeBytes.length + data.length);\n    result.set(idBytes, 0);\n    result.set(sizeBytes, idBytes.length);\n    result.set(data, idBytes.length + sizeBytes.length);\n    return result;\n  }\n\n  /**\n   * 编码 unsigned integer Element\n   */\n  private encodeUintElement(id: number, value: number): Uint8Array {\n    return this.encodeElement(id, this.encodeUint(value));\n  }\n\n  /**\n   * 编码 string Element\n   */\n  private encodeStringElement(id: number, value: string): Uint8Array {\n    return this.encodeElement(id, this.encodeString(value));\n  }\n\n  /**\n   * 编码 float32 Element\n   */\n  private encodeFloat32Element(id: number, value: number): Uint8Array {\n    return this.encodeElement(id, this.encodeFloat32(value));\n  }\n\n  /**\n   * 合并多个 Uint8Array\n   */\n  private concatArrays(arrays: Uint8Array[]): Uint8Array {\n    const totalLength = arrays.reduce((sum, arr) => sum + arr.length, 0);\n    const result = new Uint8Array(totalLength);\n    let offset = 0;\n    for (const arr of arrays) {\n      result.set(arr, offset);\n      offset += arr.length;\n    }\n    return result;\n  }\n\n  /**\n   * 构建 EBML Header\n   */\n  private buildEBMLHeader(): Uint8Array {\n    const children = this.concatArrays([\n      this.encodeUintElement(EBML_ID.EBMLVersion, 1),\n      this.encodeUintElement(EBML_ID.EBMLReadVersion, 1),\n      this.encodeUintElement(EBML_ID.EBMLMaxIDLength, 4),\n      this.encodeUintElement(EBML_ID.EBMLMaxSizeLength, 8),\n      this.encodeStringElement(EBML_ID.DocType, 'webm'),\n      // Opus tracks use CodecDelay and SeekPreRoll, which require WebM v4.\n      this.encodeUintElement(EBML_ID.DocTypeVersion, 4),\n      this.encodeUintElement(EBML_ID.DocTypeReadVersion, 2),\n    ]);\n    return this.encodeElement(EBML_ID.EBML, children);\n  }\n\n  /**\n   * 构建 SegmentInfo\n   */\n  private buildSegmentInfo(): Uint8Array {\n    const children = this.concatArrays([\n      this.encodeUintElement(EBML_ID.TimecodeScale, 1000000), // 1ms precision\n      this.encodeStringElement(EBML_ID.MuxingApp, 'xmov-avatar-sdk'),\n      this.encodeStringElement(EBML_ID.WritingApp, 'xmov-avatar-sdk'),\n    ]);\n    return this.encodeElement(EBML_ID.SegmentInfo, children);\n  }\n\n  /**\n   * 构建 Opus Audio Track\n   */\n  private buildOpusTracks(): Uint8Array {\n    // Opus CodecPrivate data (OpusHead)\n    // Version(1) + Channels(1) + PreSkip(2) + InputSampleRate(4) + OutputGain(2) + ChannelMappingFamily(1)\n    const opusHead = new Uint8Array([\n      0x4F, 0x70, 0x75, 0x73, 0x48, 0x65, 0x61, 0x64, // \"OpusHead\"\n      0x01,       // Version\n      0x01,       // Channel count (mono)\n      0x00, 0x00, // Pre-skip (0)\n      0x80, 0x3E, 0x00, 0x00, // Input sample rate (16000, little-endian)\n      0x00, 0x00, // Output gain (0)\n      0x00,       // Channel mapping family (0)\n    ]);\n\n    const audioChildren = this.concatArrays([\n      this.encodeFloat32Element(EBML_ID.SamplingFrequency, this.sampleRate),\n      this.encodeUintElement(EBML_ID.Channels, 1),\n    ]);\n\n    const trackEntryChildren = this.concatArrays([\n      this.encodeUintElement(EBML_ID.TrackNumber, 1),\n      this.encodeUintElement(EBML_ID.TrackUID, 1),\n      this.encodeUintElement(EBML_ID.TrackType, 2), // Audio\n      this.encodeUintElement(EBML_ID.FlagLacing, 0),\n      this.encodeStringElement(EBML_ID.Language, 'und'),\n      this.encodeStringElement(EBML_ID.CodecID, 'A_OPUS'),\n      this.encodeElement(EBML_ID.CodecPrivate, opusHead),\n      this.encodeUintElement(EBML_ID.CodecDelay, 0),\n      this.encodeUintElement(EBML_ID.SeekPreRoll, 80000000), // 80ms in ns\n      this.encodeElement(EBML_ID.Audio, audioChildren),\n    ]);\n\n    const trackEntry = this.encodeElement(EBML_ID.TrackEntry, trackEntryChildren);\n    return this.encodeElement(EBML_ID.Tracks, trackEntry);\n  }\n\n  /**\n   * 构建 Cluster + SimpleBlock\n   */\n  private buildCluster(timecodeMs: number, opusData: Uint8Array): Uint8Array {\n    // Timecode element\n    const timecode = this.encodeUintElement(EBML_ID.Timecode, timecodeMs);\n\n    // SimpleBlock: TrackNumber(VINT) + Timecode(2 bytes, signed, relative to cluster) + Flags(1 byte) + Data\n    const trackNumber = this.encodeVINT(1); // Track 1\n    const relativeTimecode = new Uint8Array([0x00, 0x00]); // 0ms relative\n    // WebCodecs Opus EncodedAudioChunk values are always key chunks.\n    const flags = new Uint8Array([0x80]);\n\n    const simpleBlockData = this.concatArrays([\n      trackNumber,\n      relativeTimecode,\n      flags,\n      opusData,\n    ]);\n    const simpleBlock = this.encodeElement(EBML_ID.SimpleBlock, simpleBlockData);\n\n    // Cluster = Timecode + SimpleBlock\n    const clusterChildren = this.concatArrays([timecode, simpleBlock]);\n    return this.encodeElement(EBML_ID.Cluster, clusterChildren);\n  }\n}\n","import { WebCodecOpusEncoder } from './webcodec-opus-encoder';\nimport { WebMMuxer } from './webm-muxer';\n\nexport class PcmWebMEncoder {\n  private readonly opusEncoder = new WebCodecOpusEncoder({\n    sampleRate: 16000,\n    channels: 1,\n    bitrate: 24000,\n    frameSize: 1600,\n  });\n  private readonly muxer = new WebMMuxer(16000);\n  private queue: Promise<void> = Promise.resolve();\n  private generation = 0;\n  private destroyed = false;\n  private pendingBlocks = 0;\n\n  static isSupported() {\n    return WebCodecOpusEncoder.isSupported();\n  }\n\n  async init() {\n    await this.opusEncoder.init();\n  }\n\n  async encode(pcm: Int16Array) {\n    if (pcm.length !== 1600) {\n      throw new Error(`PCM 音频块必须为 1600 samples，实际为 ${pcm.length}`);\n    }\n    if (this.pendingBlocks >= 20) {\n      throw new Error(\"PCM/Opus 编码队列超过 2 秒\");\n    }\n    this.pendingBlocks += 1;\n    const generation = this.generation;\n    const pcmCopy = new Int16Array(pcm);\n    const operation = this.queue.then(async () => {\n      if (this.destroyed || generation !== this.generation) {\n        return new ArrayBuffer(0);\n      }\n      const frames = await this.opusEncoder.encode(pcmCopy);\n      if (this.destroyed || generation !== this.generation) {\n        return new ArrayBuffer(0);\n      }\n      for (const frame of frames) {\n        this.muxer.feedOpusFrame(frame.data, frame.durationMs);\n      }\n      return this.muxer.flush();\n    });\n    const trackedOperation = operation.finally(() => {\n      this.pendingBlocks -= 1;\n    });\n    this.queue = trackedOperation.then(() => undefined, () => undefined);\n    return trackedOperation;\n  }\n\n  reset() {\n    const generation = ++this.generation;\n    this.queue = this.queue.then(() => {\n      if (!this.destroyed && generation === this.generation) {\n        this.opusEncoder.reset();\n        this.muxer.reset();\n      }\n    });\n  }\n\n  async destroy() {\n    this.destroyed = true;\n    this.generation += 1;\n    await this.opusEncoder.destroy();\n    await this.queue;\n    this.muxer.reset();\n  }\n}\n","import type { AgentAudioMetadata, AgentAudioOptions } from \"./types\";\nimport { FixedAudioTrack } from \"./fixed-audio-track\";\nimport { DEFAULT_AGENT_AUDIO } from \"./types\";\nimport { logAudioChunkContent } from \"./audio-debug\";\nimport {\n  beginWebAudioCaptureSession,\n  endWebAudioCaptureSession,\n} from \"../utils/audio-session\";\nimport { MediaRecorderTimestamp } from \"../utils/media-recorder-timestamp\";\nimport { PcmWebMEncoder } from \"../encoding/pcm-webm-encoder\";\n\nconst WEBM_OPUS_MIME_TYPE = \"audio/webm;codecs=opus\";\nconst TARGET_SAMPLE_RATE = 16_000;\nconst TARGET_CHANNELS = 1;\nconst PCM_BLOCK_MS = 100;\nconst PCM_CAPTURE_PROCESSOR = \"agent-pcm-capture-processor\";\nconst PCM_CAPTURE_WORKLET = `\nclass AgentPcmCaptureProcessor extends AudioWorkletProcessor {\n  constructor() {\n    super();\n    this.buffer = new Float32Array(1600);\n    this.bufferIndex = 0;\n    this.sequence = 0;\n    this.port.onmessage = (event) => {\n      if (event.data?.type === 'flush') {\n        this.emitBlock(true);\n        this.port.postMessage({ type: 'flushed' });\n      }\n    };\n  }\n\n  emitBlock(padPartial = false) {\n    if (this.bufferIndex === 0 || (!padPartial && this.bufferIndex < 1600)) return;\n    const pcm = new Int16Array(1600);\n    for (let index = 0; index < this.bufferIndex; index += 1) {\n      const sample = Math.max(-1, Math.min(1, this.buffer[index]));\n      pcm[index] = sample < 0 ? sample * 0x8000 : sample * 0x7fff;\n    }\n    this.port.postMessage(\n      { type: 'frame', pcm: pcm.buffer, sequence: this.sequence++ },\n      [pcm.buffer],\n    );\n    this.buffer.fill(0);\n    this.bufferIndex = 0;\n  }\n\n  process(inputs, outputs) {\n    for (const output of outputs) {\n      for (const channel of output) channel.fill(0);\n    }\n    const channels = inputs[0];\n    if (!channels?.length || !channels[0]?.length) return true;\n    for (let frame = 0; frame < channels[0].length; frame += 1) {\n      let sample = 0;\n      for (let channel = 0; channel < channels.length; channel += 1) {\n        sample += channels[channel][frame] || 0;\n      }\n      this.buffer[this.bufferIndex++] = sample / channels.length;\n      if (this.bufferIndex === 1600) this.emitBlock();\n    }\n    return true;\n  }\n}\nregisterProcessor('${PCM_CAPTURE_PROCESSOR}', AgentPcmCaptureProcessor);\n`;\ntype AgentCodedError = Error & {\n  agentCode?: string;\n  retryable?: boolean;\n};\n\nexport interface MicrophoneControllerOptions {\n  audio?: AgentAudioOptions;\n  debugAudioChunks?: boolean;\n  /** Agent-owned input stream. When omitted, the controller requests the microphone. */\n  inputStream?: MediaStream;\n  /** Use deterministic 100ms PCM buckets with WebCodecs Opus encoding. */\n  usePcmWebCodecs?: boolean;\n  onMetadata?: (metadata: AgentAudioMetadata) => void;\n  /** WebM/Opus Blob callback; timestamp is the media-block start time. */\n  onFrame: (frame: Blob, timestamp?: number) => void;\n  onError?: (error: Error) => void;\n}\n\nexport class MicrophoneController {\n  private stream: MediaStream | null = null;\n  private pendingInputStream: MediaStream | null;\n  private recordingStream: MediaStream | null = null;\n  private audioTransformer: FixedAudioTrack | null = null;\n  private mediaRecorder: MediaRecorder | null = null;\n  private pcmAudioContext: AudioContext | null = null;\n  private pcmSource: MediaStreamAudioSourceNode | null = null;\n  private pcmWorklet: AudioWorkletNode | null = null;\n  private pcmEncoder: PcmWebMEncoder | null = null;\n  private pcmEncodeQueue: Promise<void> = Promise.resolve();\n  private pcmStartedAt = 0;\n  private pcmGeneration = 0;\n  private resolvePcmFlush: (() => void) | null = null;\n  private startGeneration = 0;\n  private starting = false;\n  private hasAudioSessionCapture = false;\n  private metadata: AgentAudioMetadata;\n  private readonly options: MicrophoneControllerOptions;\n  private readonly usesProvidedInput: boolean;\n\n  constructor(options: MicrophoneControllerOptions) {\n    this.options = options;\n    this.pendingInputStream = options.inputStream ?? null;\n    this.usesProvidedInput = Boolean(options.inputStream);\n    this.metadata = {\n      ...DEFAULT_AGENT_AUDIO,\n      chunkMs: options.audio?.chunkMs || DEFAULT_AGENT_AUDIO.chunkMs,\n    };\n  }\n\n  get isRecording() {\n    return this.starting || Boolean(this.stream) || Boolean(this.pendingInputStream);\n  }\n\n  getAudioMetadata() {\n    return this.metadata;\n  }\n\n  async start() {\n    if (this.stream || this.starting) {\n      return;\n    }\n\n    const startGeneration = ++this.startGeneration;\n    this.starting = true;\n    try {\n      const inputStream = this.pendingInputStream;\n      this.stream = inputStream ?? null;\n      this.pendingInputStream = null;\n      const mediaDevices = navigator.mediaDevices;\n      if (!this.stream && !mediaDevices?.getUserMedia) {\n        throw new Error(\"当前浏览器不支持麦克风采集\");\n      }\n\n      const MediaRecorderCtor = globalThis.MediaRecorder;\n      if (this.options.usePcmWebCodecs) {\n        if (!await PcmWebMEncoder.isSupported()) {\n          throw this.webCodecsOpusUnsupported();\n        }\n      } else if (!MediaRecorderCtor?.isTypeSupported?.(WEBM_OPUS_MIME_TYPE)) {\n        throw this.webMOpusUnsupported();\n      }\n      this.assertStartActive(startGeneration);\n\n      // A playback-only session can prevent iOS WebKit from starting a\n      // microphone track. Switch categories before requesting capture.\n      if (!this.stream) {\n        if (this.usesProvidedInput) {\n          throw inputStream ? this.cancelled() : new Error(\"ASR 输入流已释放\");\n        }\n        this.hasAudioSessionCapture = beginWebAudioCaptureSession();\n        this.stream = await mediaDevices.getUserMedia({\n          audio: {\n            sampleRate: { ideal: TARGET_SAMPLE_RATE },\n            channelCount: { ideal: TARGET_CHANNELS },\n            echoCancellation: true,\n            noiseSuppression: true,\n            autoGainControl: true,\n          },\n        });\n      }\n      this.assertStartActive(startGeneration);\n      const inputTrack = this.stream.getAudioTracks()[0];\n      if (!inputTrack) {\n        throw new Error(\"麦克风未返回音频轨道\");\n      }\n      if (this.options.usePcmWebCodecs) {\n        await this.startPcmWebMRecorder(this.stream);\n        this.assertStartActive(startGeneration);\n        this.metadata = {\n          ...this.metadata,\n          sampleRate: TARGET_SAMPLE_RATE,\n          channels: TARGET_CHANNELS,\n          chunkMs: PCM_BLOCK_MS,\n        };\n        this.options.onMetadata?.(this.metadata);\n        return;\n      }\n      const audioTransformer = new FixedAudioTrack({\n        onError: (error) => {\n          const normalized = error instanceof Error ? error : new Error(\"16 kHz mono 音频处理失败\");\n          normalized.name = \"MediaRecorderError\";\n          this.options.onError?.(normalized);\n        },\n      });\n      this.audioTransformer = audioTransformer;\n      this.recordingStream = await audioTransformer.start(inputTrack);\n      this.assertStartActive(startGeneration);\n      const outputTrack = this.recordingStream.getAudioTracks()[0];\n      const outputSettings = outputTrack?.getSettings() as\n        | (MediaTrackSettings & { sampleRate?: number; channelCount?: number })\n        | undefined;\n      this.metadata = {\n        ...this.metadata,\n        sampleRate: outputSettings?.sampleRate || TARGET_SAMPLE_RATE,\n        channels: outputSettings?.channelCount || TARGET_CHANNELS,\n      };\n      this.startWebMRecorder(MediaRecorderCtor);\n      this.options.onMetadata?.(this.metadata);\n    } catch (error) {\n      const startCancelled = startGeneration !== this.startGeneration;\n      await this.stop();\n      const normalized: AgentCodedError = startCancelled\n        ? this.cancelled()\n        : error instanceof Error\n          ? error\n          : new Error(\"麦克风启动失败\");\n      if (normalized.agentCode !== \"AUDIO_FIXED_TRACK_CANCELLED\") {\n        this.options.onError?.(normalized);\n      }\n      throw normalized;\n    } finally {\n      if (this.startGeneration === startGeneration) {\n        this.starting = false;\n      }\n    }\n  }\n\n  async stop() {\n    this.startGeneration += 1;\n    this.starting = false;\n    try {\n      const audioTransformer = this.audioTransformer;\n      try {\n        // Drain the processor first, but keep the generator track live so the\n        // recorder can capture the flushed tail before it is explicitly stopped.\n        await audioTransformer?.drain();\n      } finally {\n        await this.stopPcmWebMRecorder();\n        await this.stopMediaRecorder();\n        await audioTransformer?.stop();\n        this.audioTransformer = null;\n      }\n      this.recordingStream = null;\n\n      if (this.stream && !audioTransformer) {\n        this.stream.getTracks().forEach((track) => track.stop());\n      }\n      this.stream = null;\n      this.pendingInputStream?.getTracks().forEach((track) => track.stop());\n      this.pendingInputStream = null;\n    } finally {\n      if (this.hasAudioSessionCapture) {\n        this.hasAudioSessionCapture = false;\n        endWebAudioCaptureSession();\n      }\n    }\n  }\n\n  private startWebMRecorder(MediaRecorderCtor: typeof MediaRecorder) {\n    if (!this.recordingStream) {\n      throw new Error(\"16 kHz mono 音频流未就绪\");\n    }\n\n    let recorder: MediaRecorder;\n    try {\n      recorder = new MediaRecorderCtor(this.recordingStream, {\n        mimeType: WEBM_OPUS_MIME_TYPE,\n        audioBitsPerSecond: 24000,\n      });\n    } catch (error) {\n      if (error instanceof Error && error.name === \"NotSupportedError\") {\n        throw this.webMOpusUnsupported(error);\n      }\n      throw error;\n    }\n    const recorderTimestamp = new MediaRecorderTimestamp(this.metadata.chunkMs);\n    recorder.ondataavailable = (event) => {\n      if (event.data.size > 0) {\n        const timestamp = recorderTimestamp.resolve(event.timecode);\n        (globalThis as typeof globalThis & { avatarSDKLogger?: { log?: (...args: unknown[]) => void } })\n          .avatarSDKLogger?.log?.(\"[Agent][Microphone]\", \"audio chunk\", {\n            size: event.data.size,\n            type: event.data.type,\n            chunkMs: this.metadata.chunkMs,\n            sampleRate: this.metadata.sampleRate,\n            channels: this.metadata.channels,\n            recorderState: recorder.state,\n            timecode: event.timecode,\n          });\n        if (this.options.debugAudioChunks) {\n          void logAudioChunkContent(\"[Agent][Microphone] nearend WebM/Opus chunk\", event.data, {\n            chunkMs: this.metadata.chunkMs,\n            sampleRate: this.metadata.sampleRate,\n            channels: this.metadata.channels,\n            recorderState: recorder.state,\n            timecode: event.timecode,\n          });\n        }\n        this.options.onFrame(event.data, timestamp);\n      }\n    };\n    recorder.onerror = (event) => {\n      const error = (event as ErrorEvent).error;\n      const recorderError = new Error(error instanceof Error ? error.message : \"WebM/Opus 麦克风录制失败\");\n      recorderError.name = \"MediaRecorderError\";\n      this.options.onError?.(recorderError);\n    };\n    this.mediaRecorder = recorder;\n    try {\n      recorder.start(this.metadata.chunkMs);\n    } catch (error) {\n      if (error instanceof Error && error.name === \"NotSupportedError\") {\n        throw this.webMOpusUnsupported(error);\n      }\n      throw error;\n    }\n  }\n\n  private assertStartActive(startGeneration: number) {\n    if (startGeneration !== this.startGeneration) {\n      throw this.cancelled();\n    }\n  }\n\n  private cancelled() {\n    const error = new Error(\"麦克风启动已取消\") as AgentCodedError;\n    error.name = \"AbortError\";\n    error.agentCode = \"AUDIO_FIXED_TRACK_CANCELLED\";\n    error.retryable = true;\n    return error;\n  }\n\n  private webMOpusUnsupported(cause?: unknown) {\n    const error = new Error(\"当前浏览器不支持 audio/webm;codecs=opus\") as AgentCodedError & {\n      cause?: unknown;\n    };\n    error.name = \"NotSupportedError\";\n    error.agentCode = \"AUDIO_WEBM_OPUS_UNSUPPORTED\";\n    error.retryable = false;\n    error.cause = cause;\n    return error;\n  }\n\n  private webCodecsOpusUnsupported(cause?: unknown) {\n    const error = new Error(\"当前浏览器不支持 WebCodecs AudioEncoder Opus\") as AgentCodedError & {\n      cause?: unknown;\n    };\n    error.name = \"NotSupportedError\";\n    error.agentCode = \"AUDIO_WEBCODECS_OPUS_UNSUPPORTED\";\n    error.retryable = false;\n    error.cause = cause;\n    return error;\n  }\n\n  private async startPcmWebMRecorder(stream: MediaStream) {\n    const globals = globalThis as typeof globalThis & {\n      webkitAudioContext?: typeof AudioContext;\n    };\n    const AudioContextCtor = globalThis.AudioContext || globals.webkitAudioContext;\n    if (!AudioContextCtor || typeof globalThis.AudioWorkletNode === \"undefined\") {\n      throw this.webCodecsOpusUnsupported(new Error(\"AudioWorklet 不可用\"));\n    }\n\n    const encoder = new PcmWebMEncoder();\n    await encoder.init();\n    const context = new AudioContextCtor({ sampleRate: TARGET_SAMPLE_RATE });\n    if (context.sampleRate !== TARGET_SAMPLE_RATE) {\n      await encoder.destroy();\n      await context.close().catch(() => undefined);\n      throw this.webCodecsOpusUnsupported(\n        new Error(`浏览器无法创建 ${TARGET_SAMPLE_RATE} Hz AudioContext`),\n      );\n    }\n    this.pcmAudioContext = context;\n    this.pcmEncoder = encoder;\n\n    const workletUrl = URL.createObjectURL(\n      new Blob([PCM_CAPTURE_WORKLET], { type: \"application/javascript\" }),\n    );\n    try {\n      await context.audioWorklet.addModule(workletUrl);\n    } finally {\n      URL.revokeObjectURL(workletUrl);\n    }\n    if (context.state === \"suspended\") {\n      await context.resume();\n    }\n    if (context.state !== \"running\") {\n      throw this.audioContextSuspended();\n    }\n\n    const source = context.createMediaStreamSource(stream);\n    const worklet = new AudioWorkletNode(context, PCM_CAPTURE_PROCESSOR, {\n      numberOfInputs: 1,\n      numberOfOutputs: 1,\n      outputChannelCount: [1],\n    });\n    const generation = ++this.pcmGeneration;\n    this.pcmSource = source;\n    this.pcmWorklet = worklet;\n    this.pcmEncodeQueue = Promise.resolve();\n    this.pcmStartedAt = this.monotonicEpochNow();\n\n    worklet.port.onmessage = (event) => {\n      if (event.data?.type === \"flushed\") {\n        this.resolvePcmFlush?.();\n        this.resolvePcmFlush = null;\n        return;\n      }\n      if (event.data?.type !== \"frame\" || !(event.data.pcm instanceof ArrayBuffer)) {\n        return;\n      }\n      const timestamp = this.pcmStartedAt + Number(event.data.sequence || 0) * PCM_BLOCK_MS;\n      const pcm = new Int16Array(event.data.pcm);\n      this.pcmEncodeQueue = this.pcmEncodeQueue\n        .then(async () => {\n          const webm = await encoder.encode(pcm);\n          if (generation !== this.pcmGeneration || webm.byteLength === 0) {\n            return;\n          }\n          const frame = new Blob([webm], { type: WEBM_OPUS_MIME_TYPE });\n          if (this.options.debugAudioChunks) {\n            void logAudioChunkContent(\"[Agent][Microphone] nearend PCM/WebCodecs chunk\", frame, {\n              chunkMs: PCM_BLOCK_MS,\n              sampleRate: TARGET_SAMPLE_RATE,\n              channels: TARGET_CHANNELS,\n              sequence: event.data.sequence,\n            });\n          }\n          this.options.onFrame(frame, timestamp);\n        })\n        .catch((cause) => {\n          const error = cause instanceof Error ? cause : new Error(\"PCM/Opus 麦克风编码失败\");\n          error.name = \"AudioEncoderError\";\n          this.options.onError?.(error);\n        });\n    };\n    source.connect(worklet);\n    worklet.connect(context.destination);\n  }\n\n  private async stopPcmWebMRecorder() {\n    const worklet = this.pcmWorklet;\n    const source = this.pcmSource;\n    const context = this.pcmAudioContext;\n    const encoder = this.pcmEncoder;\n    if (!worklet && !context && !encoder) {\n      return;\n    }\n\n    source?.disconnect();\n    let flushCompleted = true;\n    if (worklet) {\n      flushCompleted = await Promise.race([\n        new Promise<boolean>((resolve) => {\n          this.resolvePcmFlush = () => resolve(true);\n          worklet.port.postMessage({ type: \"flush\" });\n        }),\n        new Promise<boolean>((resolve) => globalThis.setTimeout(() => resolve(false), 100)),\n      ]);\n      if (!flushCompleted) {\n        this.resolvePcmFlush = null;\n        worklet.port.onmessage = null;\n      }\n    }\n    await this.pcmEncodeQueue;\n    this.pcmGeneration += 1;\n    this.resolvePcmFlush = null;\n    if (worklet) {\n      worklet.port.onmessage = null;\n      worklet.disconnect();\n      worklet.port.close();\n    }\n    if (context && context.state !== \"closed\") {\n      await context.close().catch(() => undefined);\n    }\n    await encoder?.destroy();\n    this.pcmWorklet = null;\n    this.pcmSource = null;\n    this.pcmAudioContext = null;\n    this.pcmEncoder = null;\n    this.pcmEncodeQueue = Promise.resolve();\n  }\n\n  private monotonicEpochNow() {\n    const clock = globalThis.performance;\n    if (clock && Number.isFinite(clock.timeOrigin)) {\n      return Math.round(clock.timeOrigin + clock.now());\n    }\n    return Date.now();\n  }\n\n  private audioContextSuspended() {\n    const error = new Error(\"AudioContext 未运行，请在用户操作后重试\") as AgentCodedError;\n    error.name = \"NotAllowedError\";\n    error.agentCode = \"AUDIO_CONTEXT_SUSPENDED\";\n    error.retryable = true;\n    return error;\n  }\n\n  private async stopMediaRecorder() {\n    const recorder = this.mediaRecorder;\n    this.mediaRecorder = null;\n    if (!recorder) {\n      return;\n    }\n\n    const shouldStop = recorder.state !== \"inactive\";\n    await new Promise<void>((resolve) => {\n      let settled = false;\n      const finish = () => {\n        if (settled) {\n          return;\n        }\n        settled = true;\n        globalThis.clearTimeout(timeoutId);\n        resolve();\n      };\n      const onError = recorder.onerror;\n      const timeoutId = globalThis.setTimeout(finish, 1000);\n      recorder.onstop = finish;\n      recorder.onerror = (event) => {\n        onError?.call(recorder, event);\n        finish();\n      };\n      if (shouldStop) {\n        try {\n          recorder.stop();\n        } catch {\n          finish();\n        }\n      }\n    });\n    recorder.ondataavailable = null;\n    recorder.onerror = null;\n    recorder.onstop = null;\n  }\n\n}\n","const AU_HEADER_BYTES = 8;\nconst AU_VERSION = 1;\nconst AUDIO_BLOCK_MS = 100;\nconst MAX_FAREND_BUCKET_SKEW = 1;\nconst MAX_PENDING_FAREND_BUCKETS = 3;\nconst NEAREND_GRACE_BUCKETS = 1;\nconst WEBM_EBML_MAGIC = new Uint8Array([0x1a, 0x45, 0xdf, 0xa3]);\n\nexport const AU_STREAM_NEAREND = 1 as const;\nexport const AU_STREAM_FAREND = 2 as const;\n\nexport type AuStreamId = typeof AU_STREAM_NEAREND | typeof AU_STREAM_FAREND;\nexport type AuPayload = Blob | ArrayBuffer | ArrayBufferView;\n\nexport interface AgentAudioBlock {\n  data: AuPayload;\n  /** Epoch timestamp for the beginning of this 100 ms media block. */\n  timestamp: number;\n  /** Whether this block starts a new WebM container stream. */\n  isFirstChunk?: boolean;\n}\n\ninterface PendingAudioBlock extends AgentAudioBlock {\n  streamId: AuStreamId;\n  tsMs: number;\n  frameKey: number;\n  dataParts?: AuPayload[];\n}\n\ninterface AgentAudioUplinkOptions {\n  sendBinary(frame: ArrayBuffer): void;\n  onError?(error: Error): void;\n}\n\nfunction logAudioUplink(event: string, details: Record<string, unknown>) {\n  const logger = (globalThis as typeof globalThis & {\n    avatarSDKLogger?: { log?: (...args: unknown[]) => void };\n  }).avatarSDKLogger;\n  logger?.log?.(\"[Agent][AudioUplink]\", event, details);\n}\n\nexport function packAuAudioFrame(\n  streamId: AuStreamId,\n  tsMs: number,\n  payload: ArrayBuffer | ArrayBufferView,\n): ArrayBuffer {\n  if (streamId !== AU_STREAM_NEAREND && streamId !== AU_STREAM_FAREND) {\n    throw new Error(`AU stream_id 无效: ${streamId}`);\n  }\n  if (!Number.isFinite(tsMs) || tsMs < 0 || tsMs > 0xffffffff) {\n    throw new Error(`AU ts_ms 无效: ${tsMs}`);\n  }\n\n  const body = payload instanceof ArrayBuffer\n    ? new Uint8Array(payload)\n    : new Uint8Array(payload.buffer, payload.byteOffset, payload.byteLength);\n  const frame = new ArrayBuffer(AU_HEADER_BYTES + body.byteLength);\n  const view = new DataView(frame);\n  view.setUint8(0, 0x41);\n  view.setUint8(1, 0x55);\n  view.setUint8(2, AU_VERSION);\n  view.setUint8(3, streamId);\n  view.setUint32(4, Math.floor(tsMs), false);\n  new Uint8Array(frame, AU_HEADER_BYTES).set(body);\n  return frame;\n}\n\nasync function toPayloadBytes(payload: AuPayload | AuPayload[]): Promise<Uint8Array> {\n  if (Array.isArray(payload)) {\n    const chunks = await Promise.all(payload.map(toPayloadBytes));\n    const merged = new Uint8Array(chunks.reduce((size, chunk) => size + chunk.byteLength, 0));\n    let offset = 0;\n    for (const chunk of chunks) {\n      merged.set(chunk, offset);\n      offset += chunk.byteLength;\n    }\n    return merged;\n  }\n  if (payload instanceof Blob) {\n    return new Uint8Array(await payload.arrayBuffer());\n  }\n  if (payload instanceof ArrayBuffer) {\n    return new Uint8Array(payload);\n  }\n  return new Uint8Array(payload.buffer, payload.byteOffset, payload.byteLength);\n}\n\n/**\n * Owns the connection-level AU timeline and serializes nearend/farend writes.\n * Nearend/farend are matched by 100 ms bucket. When independent capture clocks\n * straddle a bucket boundary, an aged nearend may consume farend from\n * the immediately preceding bucket. This keeps farend-before-nearend wire\n * order without uploading farend-only buckets or dropping WebM continuations.\n */\nexport class AgentAudioUplink {\n  private readonly options: AgentAudioUplinkOptions;\n  private connectionStartedAt: number | null = null;\n  private connectionGeneration = 0;\n  private sendChain: Promise<void> = Promise.resolve();\n  private failed = false;\n  private readonly pendingFarend = new Map<number, PendingAudioBlock[]>();\n  private readonly pendingNearend = new Map<number, PendingAudioBlock[]>();\n  private readonly closedNearendBuckets = new Set<number>();\n  private readonly lastTimestampByStream = new Map<AuStreamId, number>();\n  private readonly lastSentTimestampByStream = new Map<AuStreamId, number>();\n  private readonly pendingWebmPrefixByStream = new Map<AuStreamId, Uint8Array>();\n  private farendStreamState: \"unknown\" | \"ready\" | \"awaiting-header\" = \"unknown\";\n\n  constructor(options: AgentAudioUplinkOptions) {\n    this.options = options;\n  }\n\n  reset(connectionStartedAt: number | null = null) {\n    this.connectionGeneration += 1;\n    this.connectionStartedAt = connectionStartedAt;\n    this.pendingFarend.clear();\n    this.pendingNearend.clear();\n    this.closedNearendBuckets.clear();\n    this.lastTimestampByStream.clear();\n    this.lastSentTimestampByStream.clear();\n    this.pendingWebmPrefixByStream.clear();\n    this.farendStreamState = \"unknown\";\n    this.failed = false;\n    this.sendChain = Promise.resolve();\n  }\n\n  startInput() {\n    this.failed = false;\n    this.pendingFarend.clear();\n    this.pendingNearend.clear();\n    this.closedNearendBuckets.clear();\n    this.pendingWebmPrefixByStream.clear();\n    this.farendStreamState = \"unknown\";\n  }\n\n  clearFarend() {\n    this.pendingFarend.clear();\n    this.pendingWebmPrefixByStream.delete(AU_STREAM_FAREND);\n    this.farendStreamState = \"unknown\";\n  }\n\n  offerFarend(block: AgentAudioBlock) {\n    if (block.isFirstChunk === false && this.farendStreamState !== \"ready\") {\n      logAudioUplink(\"farend.drop\", {\n        reason: \"awaiting_webm_header\",\n        timestamp: block.timestamp,\n        isFirstChunk: false,\n      });\n      return;\n    }\n\n    const normalized = this.normalizeBlock(AU_STREAM_FAREND, block);\n    if (!normalized) {\n      if (block.isFirstChunk) {\n        this.abandonFarendStream();\n      }\n      return;\n    }\n\n    if (block.isFirstChunk) {\n      this.pendingFarend.clear();\n      this.pendingWebmPrefixByStream.delete(AU_STREAM_FAREND);\n      this.farendStreamState = \"ready\";\n    }\n\n    const bucket = this.pendingFarend.get(normalized.frameKey) ?? [];\n    bucket.push(normalized);\n    this.pendingFarend.set(normalized.frameKey, bucket);\n    logAudioUplink(\"farend.offer\", {\n      tsMs: normalized.tsMs,\n      frameKey: normalized.frameKey,\n      isFirstChunk: Boolean(normalized.isFirstChunk),\n      pendingNearend: [...this.pendingNearend.keys()],\n    });\n    this.flushOlderNearend(normalized.frameKey);\n    this.prune(normalized.frameKey);\n  }\n\n  offerNearend(block: AgentAudioBlock) {\n    const nearend = this.normalizeBlock(AU_STREAM_NEAREND, block);\n    if (!nearend || this.closedNearendBuckets.has(nearend.frameKey)) {\n      return;\n    }\n\n    const bucket = this.pendingNearend.get(nearend.frameKey) ?? [];\n    bucket.push(nearend);\n    this.pendingNearend.set(nearend.frameKey, bucket);\n    this.flushExpiredNearend(nearend.frameKey);\n    this.prune(nearend.frameKey);\n  }\n\n  drain() {\n    for (const frameKey of [...this.pendingNearend.keys()].sort((a, b) => a - b)) {\n      this.flushBucket(frameKey, true);\n    }\n    return this.sendChain;\n  }\n\n  private normalizeBlock(streamId: AuStreamId, block: AgentAudioBlock): PendingAudioBlock | null {\n    const connectionStartedAt = this.connectionStartedAt;\n    if (connectionStartedAt === null || !Number.isFinite(block.timestamp)) {\n      return null;\n    }\n\n    const relativeTimestamp = Math.max(0, Math.floor(block.timestamp - connectionStartedAt));\n    const frameKey = Math.floor(relativeTimestamp / AUDIO_BLOCK_MS);\n    const lastTimestamp = this.lastTimestampByStream.get(streamId);\n    if (lastTimestamp !== undefined && relativeTimestamp <= lastTimestamp) {\n      return null;\n    }\n    this.lastTimestampByStream.set(streamId, relativeTimestamp);\n    return { ...block, streamId, tsMs: relativeTimestamp, frameKey };\n  }\n\n  private enqueue(blocks: PendingAudioBlock[]) {\n    if (this.failed) {\n      return;\n    }\n    const generation = this.connectionGeneration;\n    this.sendChain = this.sendChain\n      .then(async () => {\n        if (this.failed) {\n          return;\n        }\n        for (const block of blocks) {\n          const bytes = await toPayloadBytes(block.dataParts ?? block.data);\n          if (\n            this.failed\n            || generation !== this.connectionGeneration\n            || this.connectionStartedAt === null\n            || bytes.byteLength === 0\n          ) {\n            continue;\n          }\n          if (block.streamId === AU_STREAM_FAREND && block.isFirstChunk) {\n            this.pendingWebmPrefixByStream.delete(AU_STREAM_FAREND);\n          }\n          const payload = this.coalesceSplitWebmMagic(block.streamId, bytes);\n          if (!payload) {\n            continue;\n          }\n          const lastSentTimestamp = this.lastSentTimestampByStream.get(block.streamId);\n          if (lastSentTimestamp !== undefined && block.tsMs <= lastSentTimestamp) {\n            continue;\n          }\n          this.options.sendBinary(packAuAudioFrame(block.streamId, block.tsMs, payload));\n          this.lastSentTimestampByStream.set(block.streamId, block.tsMs);\n          logAudioUplink(\"frame.send\", {\n            streamId: block.streamId,\n            tsMs: block.tsMs,\n            bytes: payload.byteLength,\n          });\n        }\n      })\n      .catch((error) => {\n        if (generation !== this.connectionGeneration) {\n          return;\n        }\n        this.failed = true;\n        this.pendingFarend.clear();\n        this.pendingNearend.clear();\n        const normalized = error instanceof Error ? error : new Error(\"AU 音频上行失败\");\n        this.options.onError?.(normalized);\n      });\n  }\n\n  /**\n   * Chromium may emit the first WebM byte (0x1a) as its own MediaRecorder\n   * Blob and put the remaining EBML magic in the next Blob. A one-byte AU\n   * payload is not a valid timed audio block, so keep an incomplete WebM\n   * magic prefix and prepend it to the next block from the same stream.\n   */\n  private coalesceSplitWebmMagic(streamId: AuStreamId, payload: Uint8Array) {\n    const pendingPrefix = this.pendingWebmPrefixByStream.get(streamId);\n    if (pendingPrefix) {\n      const merged = new Uint8Array(pendingPrefix.byteLength + payload.byteLength);\n      merged.set(pendingPrefix);\n      merged.set(payload, pendingPrefix.byteLength);\n      this.pendingWebmPrefixByStream.delete(streamId);\n      payload = merged;\n    }\n\n    if (\n      payload.byteLength < WEBM_EBML_MAGIC.byteLength\n      && payload.every((byte, index) => byte === WEBM_EBML_MAGIC[index])\n    ) {\n      this.pendingWebmPrefixByStream.set(streamId, payload.slice());\n      return null;\n    }\n\n    return payload;\n  }\n\n  private flushOlderNearend(currentFrameKey: number) {\n    for (const frameKey of [...this.pendingNearend.keys()].sort((a, b) => a - b)) {\n      if (frameKey >= currentFrameKey) {\n        break;\n      }\n      this.flushBucket(frameKey, true);\n    }\n  }\n\n  private flushExpiredNearend(currentFrameKey: number) {\n    this.flushOlderNearend(currentFrameKey - NEAREND_GRACE_BUCKETS);\n  }\n\n  private flushBucket(frameKey: number, allowPreviousFarend = false) {\n    const nearend = this.takeNearend(frameKey);\n    if (!nearend) {\n      return;\n    }\n    const previousFarendKey = frameKey - MAX_FAREND_BUCKET_SKEW;\n    const hasMatchingFarend = this.pendingFarend.has(frameKey)\n      || (allowPreviousFarend && this.pendingFarend.has(previousFarendKey));\n    const farend = hasMatchingFarend\n      ? this.takeFarendThrough(frameKey)\n      : null;\n    this.closedNearendBuckets.add(frameKey);\n    logAudioUplink(\"bucket.flush\", {\n      nearendFrameKey: frameKey,\n      pairedFarend: Boolean(farend),\n      pendingFarend: [...this.pendingFarend.keys()],\n    });\n    this.enqueue(farend ? [farend, nearend] : [nearend]);\n  }\n\n  private takeNearend(frameKey: number): PendingAudioBlock | null {\n    const blocks = this.pendingNearend.get(frameKey);\n    this.pendingNearend.delete(frameKey);\n    if (!blocks?.length) {\n      return null;\n    }\n\n    const first = blocks[0];\n    return {\n      ...first,\n      dataParts: blocks.flatMap((block) => block.dataParts ?? [block.data]),\n    };\n  }\n\n  private takeFarendThrough(frameKey: number): PendingAudioBlock | null {\n    const blocks: PendingAudioBlock[] = [];\n    for (const key of [...this.pendingFarend.keys()].sort((a, b) => a - b)) {\n      if (key > frameKey) {\n        break;\n      }\n      blocks.push(...this.pendingFarend.get(key)!);\n      this.pendingFarend.delete(key);\n    }\n    if (blocks.length === 0) {\n      return null;\n    }\n\n    const latest = blocks[blocks.length - 1];\n    return {\n      ...latest,\n      dataParts: blocks.map((block) => block.data),\n      isFirstChunk: blocks.some((block) => block.isFirstChunk),\n    };\n  }\n\n  private abandonFarendStream() {\n    logAudioUplink(\"farend.drop\", {\n      reason: \"stale_bucket_abandons_webm_stream\",\n      pendingFarend: [...this.pendingFarend.keys()],\n    });\n    this.pendingFarend.clear();\n    this.pendingWebmPrefixByStream.delete(AU_STREAM_FAREND);\n    this.farendStreamState = \"awaiting-header\";\n  }\n\n  private prune(currentFrameKey: number) {\n    const oldestAllowed = currentFrameKey - MAX_PENDING_FAREND_BUCKETS;\n    if ([...this.pendingFarend.keys()].some((frameKey) => frameKey < oldestAllowed)) {\n      // Dropping an arbitrary WebM fragment would corrupt every continuation\n      // after it. Abandon this container and wait for the next explicit header.\n      this.abandonFarendStream();\n    }\n    for (const frameKey of this.pendingNearend.keys()) {\n      if (frameKey < oldestAllowed) {\n        this.flushBucket(frameKey, true);\n      }\n    }\n    for (const frameKey of this.closedNearendBuckets) {\n      if (frameKey < oldestAllowed) {\n        this.closedNearendBuckets.delete(frameKey);\n      }\n    }\n  }\n}\n\nexport const AU_AUDIO_BLOCK_MS = AUDIO_BLOCK_MS;\n","import XmovAvatar from \"../index\";\nimport type { ISessionResponse } from \"../modules/ResourceManager\";\nimport type { IAudioPlaybackData, IAvatarOptions, IInitParams } from \"../types\";\nimport { AgentE2EClient } from \"./e2e-client\";\nimport { MicrophoneController } from \"./microphone\";\nimport { logAudioChunkContent } from \"./audio-debug\";\nimport { AgentAudioUplink } from \"./audio-uplink\";\nimport type { AgentAudioBlock } from \"./audio-uplink\";\nimport { DEFAULT_AGENT_AUDIO } from \"./types\";\nimport type {\n  AgentASRResult,\n  AgentASRStartOptions,\n  AgentASRState,\n  AgentClientMessage,\n  AgentConversationEvent,\n  AgentError,\n  AgentErrorDomain,\n  AgentLLMResponse,\n  AgentReconnectOptions,\n  AgentSemanticJudgeResult,\n  AgentServerEvent,\n  AgentSessionInfo,\n  AgentState,\n  XingyunAvatarAgentCallbacks,\n  XingyunAvatarAgentInitParams,\n  XingyunAvatarAgentOptions,\n} from \"./types\";\n\ninterface AgentRuntimeOptions {\n  e2eServer?: string;\n  authToken?: string;\n  audio?: XingyunAvatarAgentOptions[\"audio\"];\n  requestedEnableAec?: boolean;\n  callbacks?: XingyunAvatarAgentCallbacks;\n  webSocketCtor?: typeof WebSocket;\n  debugAudioChunks?: boolean;\n}\n\ntype AgentParentOptions = IAvatarOptions & {\n  sessionRequestData: Record<string, unknown>;\n};\n\ninterface InitialSessionAttempt {\n  sessionInfo: ISessionResponse;\n  e2ePromise: Promise<void>;\n  ttsaReadyPromise: Promise<void>;\n  resolveTtsaReady: () => void;\n}\n\nfunction hasSerializableConfig(config: Record<string, unknown> | undefined) {\n  if (config === undefined) {\n    return false;\n  }\n  const serializedConfig = JSON.stringify(config);\n  return serializedConfig !== undefined && serializedConfig !== \"{}\";\n}\n\nconst CLOSE_CODE_MESSAGES: Record<number, string> = {\n  1006: \"E2E WebSocket 异常断开：未收到 Close frame，请检查 E2EMPServer 日志或中间网络\",\n  4000: \"E2E WebSocket 路径未知\",\n  4001: \"E2E Token 无效\",\n  4002: \"E2E Session 不存在\",\n  4009: \"E2E 配额超限\",\n  4010: \"E2E 服务端主动断开\",\n};\n\nconst RETRYABLE_CLOSE_CODES = new Set([1001, 1006, 1011, 1012, 1013, 1014]);\nconst DEFAULT_RECONNECT_OPTIONS: Required<AgentReconnectOptions> = {\n  enabled: true,\n  maxAttempts: 6,\n  initialDelayMs: 500,\n  maxDelayMs: 8000,\n};\n\nfunction validateAgentConfigSource(\n  label: \"ASR\" | \"TTS\" | \"Brain\",\n  id: number | undefined,\n  config: Record<string, unknown> | undefined,\n) {\n  if (config !== undefined && (!config || Array.isArray(config) || typeof config !== \"object\")) {\n    throw new Error(`Agent ${label} JSON Config 必须是对象`);\n  }\n  if (id !== undefined && hasSerializableConfig(config)) {\n    throw new Error(`Agent ${label} 配置不能同时提供 ID 和 JSON Config`);\n  }\n}\n\nfunction normalizeBrainConfig(config: XingyunAvatarAgentOptions[\"brain_config\"]) {\n  if (!config || !hasSerializableConfig(config)) {\n    return undefined;\n  }\n  const extraBody = config.extra_body;\n  if (extraBody !== undefined && (!extraBody || Array.isArray(extraBody) || typeof extraBody !== \"object\")) {\n    throw new Error(\"Agent Brain extra_body 必须是 JSON 对象\");\n  }\n  return {\n    ...config,\n  };\n}\n\nfunction validateAgentObjectConfig(\n  label: \"features\" | \"extras\",\n  config: Record<string, unknown> | undefined,\n) {\n  if (config !== undefined && (!config || Array.isArray(config) || typeof config !== \"object\")) {\n    throw new Error(`Agent ${label} 必须是对象`);\n  }\n}\n\nexport default class XingyunAvatarAgent extends XmovAvatar {\n  private e2eClient: AgentE2EClient | null = null;\n  private microphone: MicrophoneController | null = null;\n  private stopASRPromise: Promise<void> | null = null;\n  private initializationPromise: Promise<ISessionResponse | null | undefined> | null = null;\n  private rejectInitialization: ((error: Error) => void) | null = null;\n  private initialSessionAttempt: InitialSessionAttempt | null = null;\n  private initialSessionGeneration = 0;\n  private readonly initialTtsaReadySessionIds = new Set<string>();\n  private initialTtsaReadyBeforeSession = false;\n  private asrStartGeneration = 0;\n  private agentState: AgentState = \"idle\";\n  private asrState: AgentASRState = \"idle\";\n  private agentDestroyed = false;\n  private agentDestroying = false;\n  private suppressNextVoiceEnd = false;\n  private audioInputEnabled: boolean;\n  private farendAudioEnabled = false;\n  private readonly audioUplink: AgentAudioUplink;\n  private originalOnAudioPlaybackData?: IAvatarOptions[\"onAudioPlaybackData\"];\n  private playbackDataHandler?: IAvatarOptions[\"onAudioPlaybackData\"];\n  private readonly agentOptions: AgentRuntimeOptions;\n  private readonly reconnectOptions: Required<AgentReconnectOptions>;\n  private reconnectTimer: number | null = null;\n  private reconnectAttempt = 0;\n  private reconnectInFlight = false;\n  private sessionReloadFallback = false;\n  private waitingForTtsaReload = false;\n  private stateBeforeReconnect: AgentState = \"ready\";\n  private reportingInitializationFailure = false;\n  private readonly handleOnlineForReconnect = () => {\n    if (this.agentState !== \"reconnecting\" || this.reconnectInFlight || this.sessionReloadFallback) {\n      return;\n    }\n    this.clearReconnectTimer();\n    this.scheduleE2EReconnect(true);\n  };\n\n  constructor(options: XingyunAvatarAgentOptions) {\n    validateAgentConfigSource(\"ASR\", options.asr_id, options.asr_config);\n    validateAgentConfigSource(\"TTS\", undefined, options.tts_config);\n    validateAgentConfigSource(\"Brain\", options.llm_id, options.brain_config);\n    validateAgentObjectConfig(\"features\", options.features);\n    validateAgentObjectConfig(\"extras\", options.extras);\n    const {\n      e2eServer,\n      authToken,\n      audio,\n      reconnect,\n      agentCallbacks,\n      webSocketCtor,\n      asr_id: asrId,\n      asr_config: asrConfig,\n      tts_config: ttsConfig,\n      features,\n      extras,\n      llm_id: llmId,\n      brain_config: brainConfig,\n      session_speak_req_id: configuredSessionSpeakReqId,\n      ...avatarOptions\n    } = options;\n    const originalOnMessage = avatarOptions.onMessage;\n    const originalOnRenderChange = avatarOptions.onRenderChange;\n    const originalOnNetworkInfo = avatarOptions.onNetworkInfo;\n    const originalOnSpeakStateChange = avatarOptions.onSpeakStateChange;\n    const originalOnVoiceStateChange = avatarOptions.onVoiceStateChange;\n    const originalOnAudioPlaybackData = avatarOptions.onAudioPlaybackData;\n    const playbackDataHandler = (data: IAudioPlaybackData) => {\n      instance?.forwardFarendAudio(data);\n      originalOnAudioPlaybackData?.(data);\n    };\n    const sessionSpeakReqId = configuredSessionSpeakReqId ?? 1;\n    const normalizedBrainConfig = normalizeBrainConfig(brainConfig);\n    let instance: XingyunAvatarAgent | undefined;\n\n    const parentOptions: AgentParentOptions = {\n      ...avatarOptions,\n      session_speak_req_id: sessionSpeakReqId,\n      sessionRequestData: {\n        ...(asrId !== undefined ? { asr_id: asrId } : {}),\n        ...(hasSerializableConfig(asrConfig) ? { asr_config: asrConfig } : {}),\n        ...(hasSerializableConfig(ttsConfig) ? { tts_config: ttsConfig } : {}),\n        ...(hasSerializableConfig(features) ? { features } : {}),\n        ...(hasSerializableConfig(extras) ? { extras } : {}),\n        ...(llmId !== undefined ? { llm_id: llmId } : {}),\n        ...(normalizedBrainConfig !== undefined\n          ? { brain_config: normalizedBrainConfig }\n          : {}),\n        session_speak_req_id: sessionSpeakReqId,\n      },\n      enableClientInterrupt: avatarOptions.enableClientInterrupt ?? true,\n      onMessage: (error) => {\n        originalOnMessage(error);\n        if (!instance?.reportingInitializationFailure) {\n          instance?.emitError(\"sdk\", String(error.code || \"AVATAR_ERROR\"), error.message || \"Avatar SDK 错误\", true, error);\n        }\n      },\n      onRenderChange: (state) => {\n        originalOnRenderChange?.(state);\n        instance?.safeEmit(() => agentCallbacks?.onRenderChange?.(state));\n      },\n      onNetworkInfo: (info) => {\n        originalOnNetworkInfo?.(info);\n        instance?.safeEmit(() => agentCallbacks?.onNetworkInfo?.(info));\n      },\n      onSpeakStateChange: (state, clientSpeakId) => {\n        originalOnSpeakStateChange?.(state, clientSpeakId);\n        instance?.safeEmit(() => agentCallbacks?.onSpeakStateChange?.({ state, clientSpeakId }));\n        if ([\"speak_start\", \"start\", \"started\"].includes(state)) {\n          instance?.emitConversation({ state: \"speaking\" });\n        }\n        if ([\"speak_end\", \"end\", \"ended\", \"completed\", \"finish\", \"finished\"].includes(state)) {\n          instance?.emitConversation({ state: \"completed\" });\n        }\n      },\n      onVoiceStateChange: (state, duration, clientSpeakId) => {\n        if (state === \"end\") {\n          instance?.forwardVoiceEnd();\n        }\n        originalOnVoiceStateChange?.(state, duration, clientSpeakId);\n      },\n      ...(originalOnAudioPlaybackData\n        ? {\n            onAudioPlaybackData: playbackDataHandler,\n          }\n        : {}),\n    };\n\n    super(parentOptions);\n    instance = this;\n    this.agentOptions = {\n      e2eServer,\n      authToken,\n      audio,\n      requestedEnableAec: typeof features?.speech_frontend?.enable_aec === \"boolean\"\n        ? features.speech_frontend.enable_aec\n        : undefined,\n      callbacks: agentCallbacks,\n      webSocketCtor,\n      debugAudioChunks: Boolean(avatarOptions.enableDebugger),\n    };\n    this.audioInputEnabled = audio?.inputEnabled ?? false;\n    this.originalOnAudioPlaybackData = originalOnAudioPlaybackData;\n    this.playbackDataHandler = playbackDataHandler;\n    this.reconnectOptions = this.normalizeReconnectOptions(reconnect);\n    this.audioUplink = new AgentAudioUplink({\n      sendBinary: (frame) => this.e2eClient?.sendBinary(frame),\n      onError: (error) => this.handleAudioUplinkError(error),\n    });\n  }\n\n  getAgentState() {\n    return this.agentState;\n  }\n\n  // Kept for callers that adopted the first Agent preview.\n  getState() {\n    return this.getAgentState();\n  }\n\n  getASRState() {\n    return this.asrState;\n  }\n\n  override init(params: XingyunAvatarAgentInitParams = {}): Promise<ISessionResponse | null | undefined> {\n    if (this.initializationPromise) {\n      return this.initializationPromise;\n    }\n    const initializationPromise = this.initialize(params).finally(() => {\n      if (this.initializationPromise === initializationPromise) {\n        this.initializationPromise = null;\n      }\n    });\n    this.initializationPromise = initializationPromise;\n    return initializationPromise;\n  }\n\n  private async initialize(params: XingyunAvatarAgentInitParams): Promise<ISessionResponse | null | undefined> {\n    this.assertNotDestroyed();\n    this.setAgentState(\"initializing\");\n    const initializationAbortPromise = new Promise<never>((_, reject) => {\n      this.rejectInitialization = reject;\n    });\n    void initializationAbortPromise.catch(() => {});\n\n    try {\n      const initParams: IInitParams = {\n        ...params,\n        onDownloadProgress: params.onDownloadProgress || (() => {}),\n      };\n      const sessionInfo = await super.init(initParams);\n      this.assertNotDestroyed();\n      if (!sessionInfo) {\n        throw new Error(\"XmovAvatar 初始化未返回 sessionInfo\");\n      }\n      this.applySessionFeatures(sessionInfo);\n\n      this.assertNotDestroyed();\n      this.beginInitialSession(sessionInfo);\n      while (true) {\n        const attempt = this.initialSessionAttempt;\n        if (!attempt) {\n          throw new Error(\"Agent 初始化缺少会话连接任务\");\n        }\n        await Promise.race([\n          Promise.all([attempt.e2ePromise, attempt.ttsaReadyPromise]),\n          initializationAbortPromise,\n        ]);\n        if (attempt === this.initialSessionAttempt) {\n          break;\n        }\n      }\n      this.assertNotDestroyed();\n      this.assertE2EOpen();\n      const initializedSessionInfo = this.initialSessionAttempt?.sessionInfo || sessionInfo;\n      this.setAgentState(\"ready\");\n      this.start();\n      return initializedSessionInfo;\n    } catch (error) {\n      if (!this.agentDestroyed && !this.agentDestroying) {\n        this.e2eClient?.disconnect(1000, \"agent_init_failed\");\n        this.e2eClient = null;\n        await this.cleanupBeforeInitComplete(\"agent_init_failed\");\n        try {\n          this.setAgentState(\"failed\");\n        } catch {\n          // Initialization failure must preserve the internal terminal state.\n        }\n        this.reportingInitializationFailure = true;\n        try {\n          this.reportInitializationFailure(\"Agent 初始化失败\", error);\n        } catch {\n          // A legacy callback must not replace the original initialization error.\n        } finally {\n          this.reportingInitializationFailure = false;\n        }\n        try {\n          this.emitError(\"sdk\", \"AGENT_INIT_FAILED\", \"Agent 初始化失败\", true, error);\n        } catch {\n          // A business callback must not interrupt initialization cleanup.\n        }\n        this.agentDestroyed = true;\n      }\n      throw error;\n    } finally {\n      this.rejectInitialization = null;\n      this.initialSessionGeneration += 1;\n      this.initialSessionAttempt?.resolveTtsaReady();\n      this.initialSessionAttempt = null;\n      this.initialTtsaReadySessionIds.clear();\n      this.initialTtsaReadyBeforeSession = false;\n    }\n  }\n\n  private beginInitialSession(sessionInfo: ISessionResponse) {\n    const previousAttempt = this.initialSessionAttempt;\n    const generation = ++this.initialSessionGeneration;\n    previousAttempt?.resolveTtsaReady();\n\n    let resolveTtsaReady = () => {};\n    const ttsaReadyPromise = new Promise<void>((resolve) => {\n      resolveTtsaReady = resolve;\n    });\n    const e2ePromise = this.replaceE2EConnection(sessionInfo.e2e_resp).catch((error) => {\n      if (generation !== this.initialSessionGeneration) {\n        return;\n      }\n      throw error;\n    });\n    const attempt: InitialSessionAttempt = {\n      sessionInfo,\n      e2ePromise,\n      ttsaReadyPromise,\n      resolveTtsaReady,\n    };\n    this.initialSessionAttempt = attempt;\n\n    if (this.initialTtsaReadySessionIds.delete(sessionInfo.session_id)) {\n      attempt.resolveTtsaReady();\n    } else if (!previousAttempt && this.initialTtsaReadyBeforeSession) {\n      this.initialTtsaReadyBeforeSession = false;\n      attempt.resolveTtsaReady();\n    }\n\n    void e2ePromise.catch(() => {});\n    return e2ePromise;\n  }\n\n  override start() {\n    if (this.agentState === \"initializing\") {\n      return;\n    }\n    if (this.agentState === \"running\") {\n      return;\n    }\n    if (this.agentState === \"reconnecting\") {\n      if (\n        this.waitingForTtsaReload\n        && this.e2eClient?.isOpen\n      ) {\n        if (this.stateBeforeReconnect === \"running\") {\n          super.start();\n        }\n        this.finishReconnect();\n      }\n      return;\n    }\n    this.assertE2EOpen();\n    super.start();\n    this.setAgentState(\"running\");\n  }\n\n  protected override onTtsaReady(sessionInfo?: ISessionResponse) {\n    if (this.rejectInitialization) {\n      const attempt = this.initialSessionAttempt;\n      const sessionId = sessionInfo?.session_id;\n      if (attempt && (!sessionId || attempt.sessionInfo.session_id === sessionId)) {\n        attempt.resolveTtsaReady();\n      } else if (sessionId) {\n        this.initialTtsaReadySessionIds.add(sessionId);\n      } else {\n        this.initialTtsaReadyBeforeSession = true;\n      }\n      return;\n    }\n    if (this.agentState === \"reconnecting\") {\n      this.start();\n      return;\n    }\n    if (this.agentState === \"running\") {\n      super.start();\n    }\n  }\n\n  private abortInitialization(error: Error) {\n    this.rejectInitialization?.(error);\n  }\n\n  override async stop() {\n    if (this.agentDestroyed) {\n      return;\n    }\n    const wasReconnecting = this.agentState === \"reconnecting\";\n    if (wasReconnecting) {\n      this.stateBeforeReconnect = \"stopped\";\n    }\n    await this.stopASR();\n    await super.stop();\n    if (!wasReconnecting) {\n      this.setAgentState(\"stopped\");\n    }\n  }\n\n  override speak(...args: Parameters<XmovAvatar[\"speak\"]>) {\n    if (args[2] ?? true) {\n      this.suppressNextVoiceEnd = true;\n    }\n    return super.speak(...args);\n  }\n\n  override async destroy(reason = \"user\") {\n    if (this.agentDestroyed) {\n      if (this.agentState !== \"destroyed\") {\n        this.setAgentState(\"destroyed\");\n      }\n      return;\n    }\n    if (this.agentDestroying) {\n      return;\n    }\n    this.agentDestroying = true;\n    this.abortInitialization(new Error(\"Agent 初始化已取消\"));\n    this.cancelReconnect();\n    this.cancelSessionRestart();\n    try {\n      await this.stopASR();\n    } catch (error) {\n      this.emitError(\"sdk\", \"AGENT_ASR_STOP_FAILED\", \"Agent 销毁时停止 ASR 失败\", false, error);\n    }\n\n    this.e2eClient?.disconnect(1000, reason);\n    this.e2eClient = null;\n    this.audioUplink.reset();\n\n    try {\n      await super.destroy(reason);\n    } finally {\n      this.agentDestroyed = true;\n      this.agentDestroying = false;\n      this.setAgentState(\"destroyed\");\n    }\n  }\n\n  async ask(text: string): Promise<void> {\n    this.assertRunning();\n    this.suppressNextVoiceEnd = false;\n    this.emitConversation({ state: \"asking\", text });\n    this.sendControl({\n      type: \"ask\",\n      message: { text },\n    });\n  }\n\n  async speakByE2E(text: string): Promise<void> {\n    this.assertRunning();\n    this.suppressNextVoiceEnd = false;\n    this.emitConversation({ state: \"speaking-directly\", text });\n    this.sendControl({\n      type: \"speak\",\n      message: {\n        text,\n        is_start: true,\n        is_end: true,\n      },\n    });\n  }\n\n  async startASR(options: AgentASRStartOptions = {}): Promise<void> {\n    this.assertRunning();\n    this.suppressNextVoiceEnd = false;\n    if (this.stopASRPromise) {\n      await this.stopASRPromise;\n      this.assertRunning();\n    }\n    if (this.microphone?.isRecording) {\n      return;\n    }\n    this.assertAudioChunkDuration();\n    const inputStream = options.inputStream\n      ? this.cloneASRInputStream(options.inputStream)\n      : undefined;\n\n    const asrStartGeneration = ++this.asrStartGeneration;\n    const restoreDisabledStateOnFailure = !this.audioInputEnabled;\n    const microphone = new MicrophoneController({\n      audio: this.agentOptions.audio,\n      debugAudioChunks: this.agentOptions.debugAudioChunks,\n      inputStream,\n      usePcmWebCodecs: this.farendAudioEnabled,\n      onFrame: (data, timestamp) => {\n        if (this.agentState === \"running\" && this.e2eClient?.isOpen) {\n          try {\n            this.sendAudioFrame({\n              data,\n              timestamp: timestamp ?? Date.now() - (\n                this.agentOptions.audio?.chunkMs || DEFAULT_AGENT_AUDIO.chunkMs\n              ),\n            });\n          } catch {\n            // A close can race the isOpen check. Do not let the recorder\n            // callback escape, and do not send control events on a broken stream.\n            this.stopASRAfterSocketClose();\n          }\n        }\n      },\n      onError: (error) => this.handleMicrophoneError(microphone, error),\n    });\n    this.microphone = microphone;\n\n    try {\n      this.audioUplink.startInput();\n      if (restoreDisabledStateOnFailure) {\n        this.enableAudioInput();\n      }\n      this.emitConversation({ state: \"asking\" });\n      this.assertASRStartActive(asrStartGeneration);\n      this.setASRState(\"requesting-permission\");\n      this.assertASRStartActive(asrStartGeneration);\n      this.setASRState(\"starting\");\n      this.assertASRStartActive(asrStartGeneration);\n      await microphone.start();\n      this.assertRunning();\n      if (this.farendAudioEnabled) {\n        this.restartAudioPlaybackCapture();\n      }\n      this.setASRState(\"listening\");\n    } catch (error) {\n      await microphone.stop();\n      const isCancelled = (error as { agentCode?: string })?.agentCode === \"AUDIO_FIXED_TRACK_CANCELLED\";\n      const isCurrentStart = asrStartGeneration === this.asrStartGeneration;\n      const connectionUsable = !this.agentDestroyed\n        && !this.agentDestroying\n        && this.agentState === \"running\"\n        && Boolean(this.e2eClient?.isOpen);\n      if (this.microphone === microphone) {\n        this.microphone = null;\n      }\n      if (isCurrentStart && !isCancelled && restoreDisabledStateOnFailure && this.e2eClient?.isOpen) {\n        try {\n          this.disableAudioInput();\n        } catch {\n          // sendControl has already reported the transport failure.\n        }\n      }\n      if (isCurrentStart && !this.agentDestroyed && !this.agentDestroying) {\n        if (isCancelled && connectionUsable) {\n          this.setASRState(\"idle\");\n        } else if (!isCancelled) {\n          this.setASRState(\"failed\");\n        }\n      }\n      if (isCancelled && !connectionUsable) {\n        this.assertRunning();\n      }\n      throw error;\n    }\n  }\n\n  private cloneASRInputStream(inputStream: MediaStream) {\n    const track = inputStream.getAudioTracks().find((candidate) => candidate.readyState !== \"ended\");\n    if (!track) {\n      throw new Error(\"ASR 输入流不包含有效音频轨道\");\n    }\n    return new MediaStream([track.clone()]);\n  }\n\n  private assertASRStartActive(asrStartGeneration: number) {\n    if (asrStartGeneration === this.asrStartGeneration) {\n      return;\n    }\n    const error = new Error(\"麦克风启动已取消\") as Error & {\n      agentCode?: string;\n      retryable?: boolean;\n    };\n    error.name = \"AbortError\";\n    error.agentCode = \"AUDIO_FIXED_TRACK_CANCELLED\";\n    error.retryable = true;\n    throw error;\n  }\n\n  stopASR(): Promise<void> {\n    this.asrStartGeneration += 1;\n    if (this.stopASRPromise) {\n      return this.stopASRPromise;\n    }\n\n    const microphone = this.microphone;\n    if (!microphone?.isRecording) {\n      if (this.e2eClient?.isOpen) {\n        this.disableAudioInput();\n      }\n      return Promise.resolve();\n    }\n\n    this.microphone = null;\n    this.stopASRPromise = this.finishStopASR(microphone).finally(() => {\n      this.stopASRPromise = null;\n    });\n    return this.stopASRPromise;\n  }\n\n  private async finishStopASR(microphone: MicrophoneController, notifyServer = true) {\n    this.setASRState(\"stopping\");\n    await microphone.stop();\n    await this.audioUplink.drain();\n    if (notifyServer) {\n      this.disableAudioInput();\n      this.setASRState(\"idle\");\n    } else {\n      this.setASRState(\"failed\");\n    }\n  }\n\n  async interruptConversation(reason = \"user\") {\n    if (this.agentDestroyed) {\n      return;\n    }\n\n    void reason;\n\n    this.sendControl({\n      type: \"interrupt\",\n    });\n    super.interrupt(\"speak\");\n    this.emitConversation({ state: \"interrupted\" });\n  }\n\n  protected override async onSessionReloaded(sessionInfo: ISessionResponse): Promise<void> {\n    if (this.agentDestroyed || this.agentDestroying) {\n      return;\n    }\n    if (this.initializationPromise && this.agentState === \"initializing\") {\n      this.applySessionFeatures(sessionInfo);\n      await this.beginInitialSession(sessionInfo);\n      return;\n    }\n    if (this.agentState !== \"reconnecting\") {\n      this.stateBeforeReconnect = this.readRestorableState();\n      this.setAgentState(\"reconnecting\");\n    }\n    this.clearReconnectTimer();\n    this.sessionReloadFallback = true;\n    this.applySessionFeatures(sessionInfo);\n    await this.replaceE2EConnection(sessionInfo.e2e_resp);\n    this.waitingForTtsaReload = true;\n    this.sessionReloadFallback = false;\n  }\n\n  protected override onSessionReloadExhausted(error?: unknown): void {\n    if (this.rejectInitialization) {\n      this.abortInitialization(\n        error instanceof Error ? error : new Error(\"Agent 初始化期间 TTSA Session 恢复失败\"),\n      );\n      return;\n    }\n    if (this.agentState === \"reconnecting\") {\n      this.failReconnect(error);\n    }\n  }\n\n  override reloadSuccess() {\n    super.reloadSuccess();\n  }\n\n  private async replaceE2EConnection(sessionInfo?: AgentSessionInfo) {\n    const { wsUrl, token } = this.resolveE2EConnection(sessionInfo);\n    const previousClient = this.e2eClient;\n    this.audioUplink.reset();\n    let client: AgentE2EClient;\n    client = new AgentE2EClient({\n      wsUrl,\n      token,\n      audioInputEnabled: this.audioInputEnabled,\n      WebSocketCtor: this.agentOptions.webSocketCtor,\n      onStateChange: (state) => {\n        if (this.e2eClient === client) {\n          if (state === \"open\") {\n            this.audioUplink.reset(Date.now());\n          }\n          this.safeEmit(() => this.agentOptions.callbacks?.onSocketStateChange?.(state));\n        }\n      },\n      onEvent: (event) => {\n        if (this.e2eClient === client) {\n          this.handleServerEvent(event);\n        }\n      },\n      onError: (error) => {\n        if (\n          this.e2eClient === client\n          && this.agentState !== \"initializing\"\n          && this.agentState !== \"reconnecting\"\n        ) {\n          this.emitError(\"network\", \"E2E_SOCKET_ERROR\", error.message, true, error);\n        }\n      },\n      onClose: (event) => {\n        if (this.e2eClient === client) {\n          this.handleSocketClose(event);\n        }\n      },\n    });\n    this.e2eClient = client;\n    previousClient?.disconnect(1000, \"e2e_connection_replaced\");\n    await client.connect();\n  }\n\n  private resolveE2EConnection(sessionInfo?: AgentSessionInfo) {\n    const hasResponseWsUrl = Boolean(sessionInfo?.ws_url);\n    const hasResponseToken = Boolean(sessionInfo?.e2e_token);\n    if (hasResponseWsUrl !== hasResponseToken) {\n      throw new Error(\"统一 session 响应的 e2e_resp 必须同时返回 ws_url 和 e2e_token\");\n    }\n    if (hasResponseWsUrl && hasResponseToken) {\n      return { wsUrl: sessionInfo?.ws_url as string, token: sessionInfo?.e2e_token as string };\n    }\n\n    const hasConfiguredWsUrl = Boolean(this.agentOptions.e2eServer);\n    const hasConfiguredToken = Boolean(this.agentOptions.authToken);\n    if (hasConfiguredWsUrl !== hasConfiguredToken) {\n      throw new Error(\"e2eServer 和 authToken 必须同时配置\");\n    }\n    if (!hasConfiguredWsUrl) {\n      throw new Error(\"统一 session 响应未返回 e2e_resp.ws_url，且未配置 e2eServer\");\n    }\n    return {\n      wsUrl: this.agentOptions.e2eServer as string,\n      token: this.agentOptions.authToken as string,\n    };\n  }\n\n  private sendControl(message: AgentClientMessage) {\n    try {\n      this.e2eClient?.send(message);\n    } catch (error) {\n      this.emitError(\"network\", \"E2E_SEND_FAILED\", \"E2E WebSocket 发送失败\", true, error);\n      throw error;\n    }\n  }\n\n  private assertAudioChunkDuration() {\n    const chunkMs = this.agentOptions.audio?.chunkMs || DEFAULT_AGENT_AUDIO.chunkMs;\n    if (!this.farendAudioEnabled || chunkMs === DEFAULT_AGENT_AUDIO.chunkMs) {\n      return;\n    }\n    const error = new Error(\"AU v1 音频分片固定为 100ms\") as Error & {\n      agentCode?: string;\n      retryable?: boolean;\n    };\n    error.name = \"NotSupportedError\";\n    error.agentCode = \"AUDIO_CHUNK_DURATION_UNSUPPORTED\";\n    error.retryable = false;\n    this.emitError(\"sdk\", error.agentCode, error.message, error.retryable, error);\n    throw error;\n  }\n\n  private sendAudioFrame(frame: AgentAudioBlock) {\n    if (!this.farendAudioEnabled) {\n      if (this.agentOptions.debugAudioChunks) {\n        void logAudioChunkContent(\"[Agent][E2E] nearend raw WebM/Opus chunk\", frame.data, {\n          timestamp: frame.timestamp,\n        });\n      }\n      try {\n        this.e2eClient?.sendBinary(frame.data);\n      } catch (error) {\n        this.handleAudioUplinkError(\n          error instanceof Error ? error : new Error(\"E2E nearend 音频帧发送失败\"),\n        );\n      }\n      return;\n    }\n\n    if (this.agentOptions.debugAudioChunks) {\n      void logAudioChunkContent(\"[Agent][E2E] nearend AU chunk\", frame.data, {\n        timestamp: frame.timestamp,\n        stream_id: 1,\n      });\n    }\n    this.audioUplink.offerNearend(frame);\n  }\n\n  private handleAudioUplinkError(error: Error) {\n    this.emitError(\n      \"network\",\n      \"E2E_AUDIO_SEND_FAILED\",\n      this.farendAudioEnabled ? \"E2E AU 音频帧发送失败\" : \"E2E 音频帧发送失败\",\n      true,\n      error,\n    );\n    const microphone = this.microphone;\n    if (!microphone?.isRecording || this.stopASRPromise) {\n      return;\n    }\n\n    this.setASRState(\"failed\");\n    this.microphone = null;\n    const cleanupPromise = this.finishFailedASR(microphone).finally(() => {\n      if (this.stopASRPromise === cleanupPromise) {\n        this.stopASRPromise = null;\n      }\n    });\n    this.stopASRPromise = cleanupPromise;\n  }\n\n  private applySessionFeatures(sessionInfo: ISessionResponse) {\n    const returnedEnableAec = sessionInfo.e2e_resp?.features?.speech_frontend?.enable_aec\n      ?? sessionInfo.features?.speech_frontend?.enable_aec;\n    this.farendAudioEnabled = typeof returnedEnableAec === \"boolean\"\n      ? returnedEnableAec\n      : this.agentOptions.requestedEnableAec\n        ?? this.agentOptions.audio?.echoCancellationEnabled\n        ?? false;\n\n    if (!this.farendAudioEnabled) {\n      this.audioUplink.clearFarend();\n    }\n\n    if (this.farendAudioEnabled || this.originalOnAudioPlaybackData) {\n      this.setAudioPlaybackDataHandler(this.playbackDataHandler);\n      if (this.farendAudioEnabled) {\n        void this.enableAudioPlaybackCapture();\n      }\n      return;\n    }\n    this.setAudioPlaybackDataHandler(undefined);\n  }\n\n  private forwardFarendAudio(data: IAudioPlaybackData) {\n    const microphoneRecording = Boolean(this.microphone?.isRecording);\n    const e2eOpen = Boolean(this.e2eClient?.isOpen);\n    if (!this.farendAudioEnabled || !microphoneRecording || !e2eOpen) {\n      (globalThis as typeof globalThis & {\n        avatarSDKLogger?: { log?: (...args: unknown[]) => void };\n      }).avatarSDKLogger?.log?.(\"[Agent][AudioUplink]\", \"farend.drop\", {\n        reason: !this.farendAudioEnabled\n          ? \"aec_disabled\"\n          : !microphoneRecording\n            ? \"microphone_not_recording\"\n            : \"e2e_not_open\",\n        timestamp: data.timestamp,\n        isFirstChunk: data.isFirstChunk,\n      });\n      return;\n    }\n\n    if (this.agentOptions.debugAudioChunks) {\n      void logAudioChunkContent(\"[Agent][E2E] farend AU chunk\", data.data, {\n        timestamp: data.timestamp,\n        stream_id: 2,\n        codec: data.codec,\n        sampleRate: data.sampleRate,\n        channels: 1,\n        samples: data.samples,\n        speech_id: data.speech_id,\n        isFirstChunk: data.isFirstChunk,\n      });\n    }\n    this.audioUplink.offerFarend({\n      data: data.data,\n      timestamp: data.timestamp,\n      isFirstChunk: data.isFirstChunk,\n    });\n  }\n\n  private handleMicrophoneError(microphone: MicrophoneController, error: Error) {\n    const codedError = error as Error & { agentCode?: string; retryable?: boolean };\n    const isRecorderError = error.name === \"MediaRecorderError\";\n    const isEncoderError = error.name === \"AudioEncoderError\";\n    const isAudioPipelineError = isRecorderError || isEncoderError;\n    const isCapabilityError = error.name === \"NotSupportedError\";\n    const isSDKAudioError = Boolean(codedError.agentCode);\n    const wasStopping = this.asrState === \"stopping\";\n    const shouldCleanup = isAudioPipelineError && this.microphone === microphone && !wasStopping;\n\n    if (shouldCleanup) {\n      this.microphone = null;\n      const cleanupPromise = this.finishFailedASR(microphone).finally(() => {\n        if (this.stopASRPromise === cleanupPromise) {\n          this.stopASRPromise = null;\n        }\n      });\n      this.stopASRPromise = cleanupPromise;\n    }\n\n    this.emitError(\n      isAudioPipelineError ? \"asr\" : isSDKAudioError || isCapabilityError ? \"sdk\" : \"permission\",\n      codedError.agentCode || (\n        isRecorderError\n          ? \"MEDIA_RECORDER_ERROR\"\n          : isEncoderError\n            ? \"AUDIO_ENCODER_ERROR\"\n            : \"MICROPHONE_ERROR\"\n      ),\n      error.message,\n      codedError.retryable ?? !isCapabilityError,\n      error,\n    );\n  }\n\n  private async finishFailedASR(microphone: MicrophoneController) {\n    try {\n      await microphone.stop();\n      await this.audioUplink.drain();\n    } catch (stopError) {\n      this.emitError(\"sdk\", \"MICROPHONE_CLEANUP_FAILED\", \"麦克风异常后的资源清理失败\", false, stopError);\n    }\n\n    try {\n      this.disableAudioInput();\n    } catch {\n      // sendControl has already reported the transport failure.\n    }\n  }\n\n  private enableAudioInput() {\n    if (this.audioInputEnabled) {\n      return;\n    }\n    this.sendControl({\n      type: \"event\",\n      message: \"audio_input_on\",\n    });\n    this.audioInputEnabled = true;\n    this.e2eClient?.setAudioInputEnabled(true);\n  }\n\n  private disableAudioInput() {\n    if (!this.audioInputEnabled) {\n      return;\n    }\n    this.sendControl({\n      type: \"event\",\n      message: \"audio_input_off\",\n    });\n    this.audioInputEnabled = false;\n    this.e2eClient?.setAudioInputEnabled(false);\n  }\n\n  private handleServerEvent(event: AgentServerEvent) {\n    if (this.agentDestroyed || this.agentDestroying) {\n      return;\n    }\n\n    switch (event.type) {\n      case \"pong\":\n        return;\n      case \"asr_result\":\n        this.handleASRResult(event);\n        return;\n      case \"llm_response\":\n        this.handleLLMResponse(event);\n        return;\n      case \"semantic_judge_round_result\":\n        this.handleSemanticJudgeResult(event);\n        return;\n      case \"error\":\n        this.handleBackendError(event);\n        return;\n      default:\n        return;\n    }\n  }\n\n  private handleASRResult(event: AgentServerEvent) {\n    const result: AgentASRResult = {\n      text: typeof event.text === \"string\" ? event.text : \"\",\n      isFinal: event.is_final === true,\n      raw: event,\n    };\n    this.safeEmit(() => this.agentOptions.callbacks?.onASRResult?.(result));\n  }\n\n  private handleLLMResponse(event: AgentServerEvent) {\n    if (event.event !== \"chunk\" && event.event !== \"done\") {\n      return;\n    }\n\n    const usage = event.usage;\n    const response: AgentLLMResponse = {\n      event: event.event,\n      ...(typeof event.text === \"string\" ? { text: event.text } : {}),\n      ...(event.is_first === true ? { isFirst: true } : {}),\n      ...(usage\n        ? {\n            usage: {\n              promptTokens: this.readFiniteNumber(usage.prompt_tokens),\n              completionTokens: this.readFiniteNumber(usage.completion_tokens),\n              totalTokens: this.readFiniteNumber(usage.total_tokens),\n              cachedTokens: this.readFiniteNumber(usage.cached_tokens),\n            },\n          }\n        : {}),\n      raw: event,\n    };\n    this.safeEmit(() => this.agentOptions.callbacks?.onLLMResponse?.(response));\n  }\n\n  private handleSemanticJudgeResult(event: AgentServerEvent) {\n    if (!event.message || typeof event.message !== \"object\") {\n      return;\n    }\n\n    const message = event.message as Record<string, unknown>;\n    const result: AgentSemanticJudgeResult = {\n      query: typeof message.query === \"string\" ? message.query : \"\",\n      meaningful: message.meaningful === true,\n      action: typeof message.action === \"string\" ? message.action : \"\",\n      raw: event,\n    };\n    this.safeEmit(() => this.agentOptions.callbacks?.onSemanticJudgeResult?.(result));\n  }\n\n  private forwardVoiceEnd() {\n    if (this.suppressNextVoiceEnd) {\n      this.suppressNextVoiceEnd = false;\n      return;\n    }\n    if (\n      this.agentDestroyed ||\n      this.agentDestroying ||\n      this.agentState !== \"running\" ||\n      !this.e2eClient?.isOpen\n    ) {\n      return;\n    }\n\n    try {\n      this.sendControl({ type: \"event\", message: \"voice_end\" });\n    } catch {\n      // sendControl has already reported the transport failure. A render\n      // callback must not throw back into the TTSA frame processing chain.\n    }\n  }\n\n  private readFiniteNumber(value: unknown) {\n    return typeof value === \"number\" && Number.isFinite(value) ? value : 0;\n  }\n\n  private handleBackendError(event: AgentServerEvent) {\n    const supportedDomains: AgentErrorDomain[] = [\"permission\", \"network\", \"asr\", \"brain\", \"ttsa\", \"quota\", \"sdk\"];\n    const domain = supportedDomains.includes(event.domain as AgentErrorDomain)\n      ? event.domain as AgentErrorDomain\n      : \"sdk\";\n    const code = String(event.code ?? \"BACKEND_ERROR\");\n    const message = typeof event.message === \"string\" ? event.message : \"E2E 后端错误\";\n    if (domain === \"asr\") {\n      this.stopASRAfterSocketClose();\n    }\n    this.emitConversation({ state: \"failed\" });\n    this.emitError(domain, code, message, domain !== \"quota\", event);\n  }\n\n  private handleSocketClose(event: CloseEvent) {\n    if (this.agentDestroyed || this.agentDestroying) {\n      return;\n    }\n    if (this.rejectInitialization) {\n      this.abortInitialization(\n        new Error(`Agent 初始化期间 E2E WebSocket 已关闭: ${event.code} ${event.reason || \"\"}`.trim()),\n      );\n      return;\n    }\n    this.audioUplink.reset();\n    this.stopASRAfterSocketClose();\n    if (this.shouldReconnect(event)) {\n      this.beginReconnect(event);\n      return;\n    }\n    this.setAgentState(\"failed\");\n    const domain: AgentErrorDomain = event.code === 4009 ? \"quota\" : \"network\";\n    const message = event.reason || CLOSE_CODE_MESSAGES[event.code] || \"E2E WebSocket 已关闭\";\n    this.emitError(domain, `E2E_CLOSE_${event.code}`, message, domain !== \"quota\", {\n      code: event.code,\n      reason: event.reason,\n      wasClean: event.wasClean,\n    });\n  }\n\n  private shouldReconnect(event: CloseEvent) {\n    return this.reconnectOptions.enabled\n      && this.agentState !== \"idle\"\n      && this.agentState !== \"initializing\"\n      && this.agentState !== \"destroyed\"\n      && RETRYABLE_CLOSE_CODES.has(event.code);\n  }\n\n  private beginReconnect(event: CloseEvent) {\n    if (this.agentState !== \"reconnecting\") {\n      this.stateBeforeReconnect = this.readRestorableState();\n      this.reconnectAttempt = 0;\n      this.sessionReloadFallback = false;\n      this.setAgentState(\"reconnecting\");\n      globalThis.addEventListener?.(\"online\", this.handleOnlineForReconnect);\n      this.emitError(\"network\", \"E2E_RECONNECTING\", \"E2E WebSocket 已断开，正在重连\", true, {\n        code: event.code,\n        reason: event.reason,\n        wasClean: event.wasClean,\n      });\n    }\n    this.audioInputEnabled = false;\n    this.e2eClient?.setAudioInputEnabled(false);\n    this.scheduleE2EReconnect(false);\n  }\n\n  private scheduleE2EReconnect(immediate: boolean) {\n    if (\n      this.agentDestroyed\n      || this.agentDestroying\n      || this.agentState !== \"reconnecting\"\n      || this.reconnectTimer !== null\n      || this.reconnectInFlight\n      || this.sessionReloadFallback\n    ) {\n      return;\n    }\n    if (this.reconnectAttempt >= this.reconnectOptions.maxAttempts) {\n      void this.fallbackToSessionReload();\n      return;\n    }\n\n    const delay = immediate ? 0 : this.getReconnectDelay(this.reconnectAttempt);\n    // @ts-ignore\n    this.reconnectTimer = globalThis.setTimeout(() => {\n      this.reconnectTimer = null;\n      void this.attemptE2EReconnect();\n    }, delay);\n  }\n\n  private async attemptE2EReconnect() {\n    const client = this.e2eClient;\n    if (!client || this.agentDestroyed || this.agentDestroying || this.agentState !== \"reconnecting\") {\n      return;\n    }\n\n    this.reconnectInFlight = true;\n    this.reconnectAttempt += 1;\n    try {\n      await client.connect();\n      if (this.e2eClient === client && !this.agentDestroyed && !this.agentDestroying) {\n        if (!this.waitingForTtsaReload) {\n          this.finishReconnect();\n        }\n      }\n    } catch {\n      if (this.e2eClient === client && !this.agentDestroyed && !this.agentDestroying) {\n        this.scheduleE2EReconnect(false);\n      }\n    } finally {\n      this.reconnectInFlight = false;\n      if (\n        this.e2eClient === client\n        && this.agentState === \"reconnecting\"\n        && this.reconnectTimer === null\n        && !this.sessionReloadFallback\n      ) {\n        this.scheduleE2EReconnect(false);\n      }\n    }\n  }\n\n  private async fallbackToSessionReload() {\n    if (this.sessionReloadFallback || this.agentDestroyed || this.agentDestroying) {\n      return;\n    }\n    this.sessionReloadFallback = true;\n    this.clearReconnectTimer();\n    try {\n      await this.restartSessionForTransport(\"e2e_reconnect_exhausted\");\n    } catch (error) {\n      this.failReconnect(error);\n    }\n  }\n\n  private finishReconnect() {\n    const nextState = this.stateBeforeReconnect;\n    this.cancelReconnect();\n    if (this.asrState !== \"idle\") {\n      this.setASRState(\"idle\");\n    }\n    this.setAgentState(nextState);\n  }\n\n  private failReconnect(cause?: unknown) {\n    this.cancelReconnect();\n    this.setAgentState(\"failed\");\n    this.emitConversation({ state: \"failed\" });\n    this.emitError(\n      \"network\",\n      \"E2E_RECONNECT_EXHAUSTED\",\n      \"E2E WebSocket 重连失败，请重新初始化 Agent\",\n      false,\n      cause,\n    );\n  }\n\n  private cancelReconnect() {\n    this.clearReconnectTimer();\n    globalThis.removeEventListener?.(\"online\", this.handleOnlineForReconnect);\n    this.reconnectAttempt = 0;\n    this.reconnectInFlight = false;\n    this.sessionReloadFallback = false;\n    this.waitingForTtsaReload = false;\n  }\n\n  private clearReconnectTimer() {\n    if (this.reconnectTimer !== null) {\n      globalThis.clearTimeout(this.reconnectTimer);\n      this.reconnectTimer = null;\n    }\n  }\n\n  private getReconnectDelay(attempt: number) {\n    const baseDelay = Math.min(\n      this.reconnectOptions.maxDelayMs,\n      this.reconnectOptions.initialDelayMs * (2 ** attempt),\n    );\n    if (baseDelay === 0) {\n      return 0;\n    }\n    return Math.round(baseDelay * (0.8 + Math.random() * 0.4));\n  }\n\n  private readRestorableState(): AgentState {\n    if (this.agentState === \"running\" || this.agentState === \"stopped\" || this.agentState === \"ready\") {\n      return this.agentState;\n    }\n    return \"ready\";\n  }\n\n  private normalizeReconnectOptions(options?: AgentReconnectOptions): Required<AgentReconnectOptions> {\n    const initialDelayMs = this.readNonNegativeNumber(\n      options?.initialDelayMs,\n      DEFAULT_RECONNECT_OPTIONS.initialDelayMs,\n    );\n    return {\n      enabled: options?.enabled ?? DEFAULT_RECONNECT_OPTIONS.enabled,\n      maxAttempts: Math.max(1, Math.floor(this.readNonNegativeNumber(\n        options?.maxAttempts,\n        DEFAULT_RECONNECT_OPTIONS.maxAttempts,\n      ))),\n      initialDelayMs,\n      maxDelayMs: Math.max(initialDelayMs, this.readNonNegativeNumber(\n        options?.maxDelayMs,\n        DEFAULT_RECONNECT_OPTIONS.maxDelayMs,\n      )),\n    };\n  }\n\n  private readNonNegativeNumber(value: unknown, fallback: number) {\n    return typeof value === \"number\" && Number.isFinite(value) && value >= 0 ? value : fallback;\n  }\n\n  private stopASRAfterSocketClose() {\n    this.asrStartGeneration += 1;\n    if (this.stopASRPromise) {\n      return;\n    }\n    const microphone = this.microphone;\n    if (!microphone?.isRecording) {\n      return;\n    }\n\n    this.microphone = null;\n    this.stopASRPromise = this.finishStopASR(microphone, false)\n      .catch((error) => {\n        this.emitError(\"sdk\", \"MICROPHONE_CLEANUP_FAILED\", \"WebSocket 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