@amaster.ai/asr-client 1.1.3 → 1.1.5

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Files changed (6) hide show
  1. package/README.md +1 -3
  2. package/dist/index.cjs.map +1 -1
  3. package/dist/index.d.cts +1 -1
  4. package/dist/index.d.ts +1 -1
  5. package/dist/index.js.map +1 -1
  6. package/package.json +2 -2
package/README.md CHANGED
@@ -321,9 +321,7 @@ interface ASRHttpClient {
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  ```tsx
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  import { createClient } from "@amaster.ai/client";
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- const client = createClient({
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- baseURL: "https://api.amaster.ai",
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- });
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+ const client = createClient();
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  // WebSocket ASR
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  const asrClient = client.asr({
package/dist/index.cjs.map CHANGED
@@ -1 +1 @@
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- {"version":3,"sources":["../src/index.ts","../src/asr-client.ts","../src/http-asr-client.ts"],"sourcesContent":["import createASRClient, {\n type ASRClientConfig,\n type ASRClient,\n type ASRLanguage,\n type SessionConfig,\n type TurnDetectionConfig,\n type InputAudioTranscriptionConfig,\n type ClientEvent,\n type ServerEvent,\n type SessionUpdateEvent,\n type InputAudioBufferAppendEvent,\n type InputAudioBufferCommitEvent,\n type SessionFinishEvent,\n type SessionCreatedEvent,\n type SessionUpdatedEvent,\n type TranscriptionTextEvent,\n type TranscriptionCompletedEvent,\n type SessionFinishedEvent,\n type ErrorEvent,\n} from \"./asr-client\";\n\nexport type {\n ASRClient,\n ASRClientConfig,\n ASRLanguage,\n SessionConfig,\n TurnDetectionConfig,\n InputAudioTranscriptionConfig,\n ClientEvent,\n ServerEvent,\n SessionUpdateEvent,\n InputAudioBufferAppendEvent,\n InputAudioBufferCommitEvent,\n SessionFinishEvent,\n SessionCreatedEvent,\n SessionUpdatedEvent,\n TranscriptionTextEvent,\n TranscriptionCompletedEvent,\n SessionFinishedEvent,\n ErrorEvent,\n};\n\nexport { createASRClient };\n\nimport createASRHttpClient, {\n type ASRHttpClientConfig,\n type ASRHttpClient,\n} from \"./http-asr-client\";\nexport type { ASRHttpClient, ASRHttpClientConfig };\nexport { createASRHttpClient };\n","/**\n * ASR Realtime WebSocket Client for Qwen-ASR Realtime API\n *\n * WebSocket-based real-time speech recognition for streaming transcription.\n * Follows the Qwen-ASR Realtime API protocol with proper event handling.\n *\n * @example\n * ```typescript\n * const client = createASRClient({\n * language: \"zh\",\n * enableVAD: true,\n * onReady() {\n * console.log(\"ASR connected\");\n * },\n * onTranscript(text, isFinal) {\n * console.log(isFinal ? \"[Final]\" : \"[Interim]\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * await client.connect();\n * await client.startRecording();\n * // ... stop ...\n * await client.stopRecording();\n * await client.close();\n * ```\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\n\nexport type ASRLanguage =\n | \"zh\" // 中文（普通话、四川话、闽南语、吴语）\n | \"yue\" // 粤语\n | \"en\" // 英文\n | \"ja\" // 日语\n | \"de\" // 德语\n | \"ko\" // 韩语\n | \"ru\" // 俄语\n | \"fr\" // 法语\n | \"pt\" // 葡萄牙语\n | \"ar\" // 阿拉伯语\n | \"it\" // 意大利语\n | \"es\" // 西班牙语\n | \"hi\" // 印地语\n | \"id\" // 印尼语\n | \"th\" // 泰语\n | \"tr\" // 土耳其语\n | \"uk\" // 乌克兰语\n | \"vi\" // 越南语\n | \"cs\" // 捷克语\n | \"da\" // 丹麦语\n | \"fil\" // 菲律宾语\n | \"fi\" // 芬兰语\n | \"is\" // 冰岛语\n | \"ms\" // 马来语\n | \"no\" // 挪威语\n | \"pl\" // 波兰语\n | \"sv\"; // 瑞典语\n\n// Client event types (sent to server)\ntype ClientEventType =\n | \"session.update\"\n | \"input_audio_buffer.append\"\n | \"input_audio_buffer.commit\"\n | \"session.finish\";\n\n// Server event types (received from server)\ntype ServerEventType =\n | \"session.created\"\n | \"session.updated\"\n | \"input_audio_buffer.speech_started\"\n | \"input_audio_buffer.speech_stopped\"\n | \"input_audio_buffer.committed\"\n | \"conversation.item.input_audio_transcription.text\"\n | \"conversation.item.input_audio_transcription.completed\"\n | \"session.finished\"\n | \"error\";\n\n// Base event interface\ninterface BaseEvent {\n event_id: string;\n type: ClientEventType | ServerEventType;\n}\n\n// Client events\ninterface SessionUpdateEvent extends BaseEvent {\n type: \"session.update\";\n session: SessionConfig;\n}\n\ninterface InputAudioBufferAppendEvent extends BaseEvent {\n type: \"input_audio_buffer.append\";\n audio: string;\n}\n\ninterface InputAudioBufferCommitEvent extends BaseEvent {\n type: \"input_audio_buffer.commit\";\n}\n\ninterface SessionFinishEvent extends BaseEvent {\n type: \"session.finish\";\n}\n\ntype ClientEvent =\n | SessionUpdateEvent\n | InputAudioBufferAppendEvent\n | InputAudioBufferCommitEvent\n | SessionFinishEvent;\n\n// Server events\ninterface SessionCreatedEvent extends BaseEvent {\n type: \"session.created\";\n session: { id: string };\n}\n\ninterface SessionUpdatedEvent extends BaseEvent {\n type: \"session.updated\";\n session: SessionConfig;\n}\n\ninterface SpeechStartedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_started\";\n}\n\ninterface SpeechStoppedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_stopped\";\n}\n\ninterface InputAudioBufferCommittedEvent extends BaseEvent {\n type: \"input_audio_buffer.committed\";\n}\n\ninterface TranscriptionTextEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.text\";\n text?: string;\n stash?: string;\n transcript?: string;\n}\n\ninterface TranscriptionCompletedEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.completed\";\n text?: string;\n transcript?: string;\n}\n\ninterface SessionFinishedEvent extends BaseEvent {\n type: \"session.finished\";\n}\n\ninterface ErrorEvent extends BaseEvent {\n type: \"error\";\n error: {\n message: string;\n code?: string;\n };\n}\n\ntype ServerEvent =\n | SessionCreatedEvent\n | SessionUpdatedEvent\n | SpeechStartedEvent\n | SpeechStoppedEvent\n | InputAudioBufferCommittedEvent\n | TranscriptionTextEvent\n | TranscriptionCompletedEvent\n | SessionFinishedEvent\n | ErrorEvent;\n\n// Session configuration\ninterface TurnDetectionConfig {\n type: \"server_vad\";\n /** VAD检测阈值，推荐设为 0.0，默认值 0.2，范围 [-1, 1] */\n threshold?: number;\n /** VAD断句检测阈值(ms)，推荐设为 400，默认值 800，范围 [200, 6000] */\n silence_duration_ms?: number;\n}\n\ninterface InputAudioTranscriptionConfig {\n language?: ASRLanguage;\n}\n\ninterface SessionConfig {\n input_audio_format?: \"pcm\" | \"opus\";\n sample_rate?: 16000 | 8000;\n input_audio_transcription?: InputAudioTranscriptionConfig;\n turn_detection?: TurnDetectionConfig | null;\n}\n\ninterface RealtimeRecorder {\n start(onAudio: (base64: string) => void): Promise<void>;\n stop(): Promise<void>;\n}\n\nasync function createRealtimeRecorder(): Promise<RealtimeRecorder> {\n let stream: MediaStream | null = null;\n let ctx: AudioContext | null = null;\n let source: MediaStreamAudioSourceNode | null = null;\n let processor: ScriptProcessorNode | null = null;\n\n if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {\n throw new Error(\"未找到麦克风或无权限\");\n }\n\n return {\n async start(onAudio) {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: { sampleRate: 16000, channelCount: 1, echoCancellation: true },\n });\n\n log(\"✅ 麦克风已启动\", \"success\");\n log(\"💬 请对着麦克风说话，实时识别中...\", \"success\");\n\n ctx = new AudioContext({ sampleRate: 16000 });\n source = ctx.createMediaStreamSource(stream);\n processor = ctx.createScriptProcessor(4096, 1, 1);\n\n processor.onaudioprocess = (e) => {\n const inputData = e.inputBuffer.getChannelData(0);\n const pcm = new Int16Array(inputData.length);\n for (let i = 0; i < inputData.length; i++) {\n const s = Math.max(-1, Math.min(1, inputData[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n\n const bytes = new Uint8Array(pcm.buffer);\n let binary = \"\";\n for (let i = 0; i < bytes.length; i++) {\n binary += String.fromCharCode(bytes[i] || 0);\n }\n\n onAudio(btoa(binary));\n };\n\n source.connect(processor);\n processor.connect(ctx.destination);\n },\n\n async stop() {\n stream?.getTracks().forEach((t) => t.stop());\n source?.disconnect();\n processor?.disconnect();\n if (ctx) {\n await ctx.close();\n }\n stream = null;\n ctx = null;\n source = null;\n processor = null;\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\n// Generate unique event ID\nlet eventIdCounter = 0;\nfunction generateEventId(): string {\n return `event_${Date.now()}_${++eventIdCounter}`;\n}\n\nexport interface ASRClientConfig {\n /**\n * Audio format\n * @default \"pcm\"\n */\n audioFormat?: \"pcm\" | \"opus\";\n /**\n * Sample rate in Hz\n * @default 16000\n * @description 支持 16000 和 8000。设置为 8000 时，服务端会先升采样到16000Hz再进行识别，可能引入微小延迟。\n */\n sampleRate?: 16000 | 8000;\n /**\n * Audio source language\n * @default \"zh\"\n * @description 支持多种语言，包括 zh（中文）、yue（粤语）、en（英文）、ja（日语）等\n */\n language?: ASRLanguage;\n /**\n * Enable VAD (Voice Activity Detection) mode\n * @default true\n * @description true = VAD模式（服务端自动检测语音开始/结束），false = Manual模式（客户端手动控制）\n */\n enableVAD?: boolean;\n /**\n * VAD detection threshold\n * @default 0.2\n * @description 推荐设为 0.0。取值范围 [-1, 1]。较低的阈值会提高 VAD 的灵敏度。\n */\n vadThreshold?: number;\n /**\n * VAD silence duration threshold in milliseconds\n * @default 800\n * @description 推荐设为 400。取值范围 [200, 6000]。静音持续时长超过该阈值将被认为是语句结束。\n */\n vadSilenceDurationMs?: number;\n /**\n * Get access token for WebSocket authentication\n */\n getAccessToken?: () => string | null;\n /**\n * Called when connection is ready (session.created received and session.update sent)\n */\n onReady?: () => void;\n /**\n * Called when speech is detected (VAD mode only)\n */\n onSpeechStart?: () => void;\n /**\n * Called when speech stops (VAD mode only)\n */\n onSpeechEnd?: () => void;\n /**\n * Called on transcript result\n * @param text - Transcribed text\n * @param isFinal - Whether this is the final result\n */\n onTranscript?: (text: string, isFinal: boolean) => void;\n /**\n * Called when audio buffer is committed (non-VAD mode only)\n */\n onAudioBufferCommitted?: () => void;\n /**\n * Called when session is finished\n */\n onSessionFinished?: () => void;\n /**\n * Called on error\n */\n onError?: (error: Error) => void;\n /**\n * Called on close\n */\n onClose?: () => void;\n}\n\nexport interface ASRClient {\n /** Connect to ASR service and establish session */\n connect(): Promise<void>;\n /** Start recording from microphone */\n startRecording(): Promise<void>;\n /**\n * Stop recording\n * @description In non-VAD mode, this triggers recognition by sending input_audio_buffer.commit\n */\n stopRecording(): Promise<void>;\n /**\n * Close connection gracefully\n * @description Sends session.finish and waits for session.finished before closing\n */\n close(): Promise<void>;\n /**\n * Check if currently recording\n */\n isRecording(): boolean;\n /**\n * Check if connected to server\n */\n isConnected(): boolean;\n}\n\nfunction createASRClient(config: ASRClientConfig): ASRClient {\n const {\n onReady,\n onSpeechStart,\n onSpeechEnd,\n onTranscript,\n onAudioBufferCommitted,\n onSessionFinished,\n onError,\n onClose,\n getAccessToken,\n audioFormat = \"pcm\",\n sampleRate = 16000,\n language = \"zh\",\n enableVAD = true,\n vadThreshold = 0.2,\n vadSilenceDurationMs = 400,\n } = config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\n let isRecordingFlag = false;\n let isClosing = false;\n\n const path = ASR_PATH;\n\n /**\n * Send event to server\n */\n function sendEvent(event: ClientEvent): void {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n ws.send(JSON.stringify(event));\n }\n\n /**\n * Build session configuration\n */\n function buildSessionConfig(): SessionConfig {\n const sessionConfig: SessionConfig = {\n input_audio_format: audioFormat,\n sample_rate: sampleRate,\n input_audio_transcription: {\n language,\n },\n };\n\n if (enableVAD) {\n sessionConfig.turn_detection = {\n type: \"server_vad\",\n threshold: vadThreshold,\n silence_duration_ms: vadSilenceDurationMs,\n };\n } else {\n // Non-VAD mode: set turn_detection to null\n sessionConfig.turn_detection = null;\n }\n\n return sessionConfig;\n }\n\n /**\n * Send session.update event\n */\n function sendSessionUpdate(): void {\n const event: SessionUpdateEvent = {\n event_id: generateEventId(),\n type: \"session.update\",\n session: buildSessionConfig(),\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.append event\n */\n function sendAudioBufferAppend(audio: string): void {\n const event: InputAudioBufferAppendEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.append\",\n audio,\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.commit event (for non-VAD mode)\n */\n function sendAudioBufferCommit(): void {\n const event: InputAudioBufferCommitEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.commit\",\n };\n sendEvent(event);\n }\n\n /**\n * Send session.finish event\n */\n function sendSessionFinish(): void {\n const event: SessionFinishEvent = {\n event_id: generateEventId(),\n type: \"session.finish\",\n };\n sendEvent(event);\n }\n\n /**\n * Handle server events\n */\n function handleServerEvent(data: ServerEvent): void {\n switch (data.type) {\n case \"session.created\":\n // Send session.update immediately after session.created\n try {\n sendSessionUpdate();\n } catch (err) {\n onError?.(\n new Error(\n \"Failed to send session.update: \" + (err instanceof Error ? err.message : String(err))\n )\n );\n }\n break;\n\n case \"session.updated\":\n // Server confirmed session configuration\n onReady?.();\n break;\n\n case \"input_audio_buffer.speech_started\":\n onSpeechStart?.();\n break;\n\n case \"input_audio_buffer.speech_stopped\":\n onSpeechEnd?.();\n break;\n\n case \"input_audio_buffer.committed\":\n onAudioBufferCommitted?.();\n break;\n\n case \"conversation.item.input_audio_transcription.text\":\n onTranscript?.(data.text || data.stash || data.transcript || \"\", false);\n break;\n\n case \"conversation.item.input_audio_transcription.completed\":\n onTranscript?.(data.text || data.transcript || \"\", true);\n break;\n\n case \"session.finished\":\n onSessionFinished?.();\n // Close WebSocket after receiving session.finished\n close();\n break;\n\n case \"error\":\n const err = new Error(data.error?.message || \"ASR error\");\n onError?.(err);\n break;\n\n default:\n // Unknown event type\n console.warn(\"[ASR] Unknown server event:\", (data as ServerEvent).type);\n }\n }\n\n async function connect(): Promise<void> {\n // Build WebSocket URL with optional token parameter\n let wsUrl = path;\n if (getAccessToken) {\n const token = getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n wsUrl = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n // Support both ws:// and wss:// protocols\n if (typeof window !== \"undefined\" && window.location) {\n const protocol = window.location.protocol === \"https:\" ? \"wss:\" : \"ws:\";\n if (!wsUrl.startsWith(\"ws://\") && !wsUrl.startsWith(\"wss://\")) {\n wsUrl = `${protocol}//${window.location.host}${wsUrl}`;\n }\n }\n\n ws = new WebSocket(wsUrl);\n\n return new Promise((resolve, reject) => {\n if (!ws) {\n reject(new Error(\"Failed to create WebSocket\"));\n return;\n }\n\n ws.onopen = () => {\n log(\"WebSocket connected\", \"success\");\n };\n\n ws.onmessage = (event) => {\n try {\n const data = JSON.parse(event.data) as ServerEvent;\n handleServerEvent(data);\n\n // Resolve on session.updated (ready state)\n if (data.type === \"session.updated\") {\n resolve();\n }\n } catch (err) {\n const error = new Error(\n \"Failed to parse server message: \" + (err instanceof Error ? err.message : String(err))\n );\n onError?.(error);\n reject(error);\n }\n };\n\n ws.onerror = (error) => {\n console.error(\"WebSocket error:\", error);\n const err = new Error(\"WebSocket error\");\n onError?.(err);\n reject(err);\n };\n\n ws.onclose = () => {\n isRecordingFlag = false;\n ws = null;\n onClose?.();\n };\n });\n }\n\n async function startRecording(): Promise<void> {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n\n if (isRecordingFlag) {\n return;\n }\n\n try {\n recorder = await createRealtimeRecorder();\n isRecordingFlag = true;\n\n await recorder.start((audio) => {\n if (!ws || ws.readyState !== WebSocket.OPEN) return;\n try {\n sendAudioBufferAppend(audio);\n } catch (err) {\n console.error(\"[ASR] Failed to send audio:\", err);\n }\n });\n } catch (err) {\n console.error(\"[ASR] Failed to start recorder:\", err);\n onError?.(err instanceof Error ? err : new Error(String(err)));\n throw err;\n }\n }\n\n async function stopRecording(): Promise<void> {\n if (!isRecordingFlag) {\n return;\n }\n\n try {\n await recorder?.stop();\n } catch (err) {\n console.error(\"[ASR] Error stopping recorder:\", err);\n }\n recorder = null;\n isRecordingFlag = false;\n\n // In non-VAD mode, send commit to trigger recognition\n if (!enableVAD && ws?.readyState === WebSocket.OPEN) {\n try {\n sendAudioBufferCommit();\n } catch (err) {\n console.error(\"[ASR] Failed to send commit:\", err);\n }\n }\n }\n\n async function close(): Promise<void> {\n if (isClosing) {\n return;\n }\n\n isClosing = true;\n\n // Stop recording first\n await stopRecording();\n\n // Send session.finish if connected\n if (ws?.readyState === WebSocket.OPEN) {\n try {\n sendSessionFinish();\n // Wait a bit for session.finished response\n await new Promise((resolve) => setTimeout(resolve, 1000));\n } catch (err) {\n console.error(\"[ASR] Failed to send session.finish:\", err);\n }\n }\n\n // Close WebSocket\n if (ws && ws?.readyState !== WebSocket.CLOSING && ws?.readyState !== WebSocket.CLOSED) {\n ws?.close();\n }\n ws = null;\n isClosing = false;\n }\n\n function isRecording(): boolean {\n return isRecordingFlag;\n }\n\n function isConnected(): boolean {\n return ws !== null && ws.readyState === WebSocket.OPEN;\n }\n\n return {\n connect,\n startRecording,\n stopRecording,\n close,\n isRecording,\n isConnected,\n };\n}\n\nexport default (authConfig: Pick<ASRClientConfig, \"getAccessToken\">) => (config: ASRClientConfig) =>\n createASRClient({ ...authConfig, ...config });\n\nexport type {\n SessionConfig,\n TurnDetectionConfig,\n InputAudioTranscriptionConfig,\n ClientEvent,\n ServerEvent,\n SessionUpdateEvent,\n InputAudioBufferAppendEvent,\n InputAudioBufferCommitEvent,\n SessionFinishEvent,\n SessionCreatedEvent,\n SessionUpdatedEvent,\n TranscriptionTextEvent,\n TranscriptionCompletedEvent,\n SessionFinishedEvent,\n ErrorEvent,\n};\n","/**\n * HTTP ASR Client - Press-to-talk style speech recognition\n *\n * HTTP-based speech recognition suitable for press-to-talk scenarios where you hold to speak\n * and release to recognize. Good for voice messages, voice search, etc.\n *\n * @example\n * ```typescript\n * const client = createASRHttpClient({\n * onRecordingStart() {\n * console.log(\"Recording started\");\n * },\n * onRecordingStop() {\n * console.log(\"Recording stopped\");\n * },\n * onResult(text) {\n * console.log(\"Recognized:\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * // Hold to speak, release to recognize\n * await client.startRecording();\n * // ... stop ...\n * const result = await client.stopRecording();\n * ```\n */\n\nimport { createHttpClient, type HttpClient } from \"@amaster.ai/http-client\";\n\nconst ASR_HTTP_PATH =\n \"/api/proxy/builtin/platform/qwen-asr/compatible-mode/v1/chat/completions\";\n\n// 录音内核\nconst RECORDER_WORKLET = `\nclass RecorderProcessor extends AudioWorkletProcessor {\n process(inputs) {\n const input = inputs[0];\n if (input && input[0]) {\n this.port.postMessage(input[0].slice(0));\n }\n return true;\n }\n}\nregisterProcessor('recorder-processor', RecorderProcessor);\n`;\n\ntype Status = \"idle\" | \"recording\" | \"recognizing\";\n\ninterface Recorder {\n /** Start recording */\n start(): Promise<void>;\n /**\n * Stop recording and get base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @returns Base64-encoded WAV audio data\n */\n stop(): Promise<void>;\n}\n\ninterface RecorderOptions {\n /** Called when recording starts */\n onStart?: () => void;\n /**\n * Called when recording stops, with base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @param base64 - Base64-encoded WAV audio data\n * @returns void\n */\n onStop?: (base64: string) => void;\n onError?: (error: Error) => void;\n}\n\n// 创建录音器\nasync function createWebRecorder(props?: RecorderOptions): Promise<Recorder> {\n let stream: MediaStream;\n let ctx: AudioContext;\n let node: AudioWorkletNode;\n let source: MediaStreamAudioSourceNode;\n const chunks: Int16Array[] = [];\n\n const cleanup = () => {\n try {\n source?.disconnect();\n node?.disconnect();\n stream?.getTracks().forEach((t) => t.stop());\n ctx?.close();\n } catch (e) {}\n };\n\n return {\n async start() {\n try {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: {\n channelCount: 1,\n echoCancellation: true,\n noiseSuppression: true,\n autoGainControl: true,\n },\n });\n\n ctx = new AudioContext();\n\n const blob = new Blob([RECORDER_WORKLET], {\n type: \"application/javascript\",\n });\n const url = URL.createObjectURL(blob);\n await ctx.audioWorklet.addModule(url);\n URL.revokeObjectURL(url);\n\n source = ctx.createMediaStreamSource(stream);\n node = new AudioWorkletNode(ctx, \"recorder-processor\");\n\n node.port.onmessage = (e) => {\n const input = e.data as Float32Array;\n const pcm = new Int16Array(input.length);\n for (let i = 0; i < input.length; i++) {\n const s = Math.max(-1, Math.min(1, input[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n chunks.push(pcm);\n };\n\n source.connect(node);\n\n props?.onStart?.();\n } catch (error) {\n props?.onError?.(\n error instanceof Error ? error : new Error(String(error)),\n );\n cleanup();\n }\n },\n\n async stop() {\n cleanup();\n\n const total = chunks.reduce((s, c) => s + c.length, 0);\n const pcm = new Int16Array(total);\n let offset = 0;\n for (const c of chunks) {\n pcm.set(c, offset);\n offset += c.length;\n }\n\n const result = { pcm, sampleRate: ctx?.sampleRate ?? 16000 };\n const base64 = await blobToBase64(\n pcmToWav(result.pcm, result.sampleRate),\n );\n props?.onStop?.(base64);\n },\n };\n}\n\n/**\n * Convert PCM to WAV Blob\n * @param pcm - PCM data\n * @param sampleRate - Sample rate\n * @returns WAV Blob\n */\nfunction pcmToWav(pcm: Int16Array, sampleRate: number): Blob {\n const buffer = new ArrayBuffer(44 + pcm.length * 2);\n const view = new DataView(buffer);\n\n const write = (o: number, s: string) => {\n for (let i = 0; i < s.length; i++) view.setUint8(o + i, s.charCodeAt(i));\n };\n\n write(0, \"RIFF\");\n view.setUint32(4, 36 + pcm.length * 2, true);\n write(8, \"WAVE\");\n write(12, \"fmt \");\n view.setUint32(16, 16, true);\n view.setUint16(20, 1, true);\n view.setUint16(22, 1, true);\n view.setUint32(24, sampleRate, true);\n view.setUint32(28, sampleRate * 2, true);\n view.setUint16(32, 2, true);\n view.setUint16(34, 16, true);\n write(36, \"data\");\n view.setUint32(40, pcm.length * 2, true);\n\n for (let i = 0; i < pcm.length; i++) {\n view.setInt16(44 + i * 2, pcm[i] || 0, true);\n }\n\n 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+ {"version":3,"sources":["../src/index.ts","../src/asr-client.ts","../src/http-asr-client.ts"],"sourcesContent":["import createASRClient, {\n type ASRClientConfig,\n type ASRClient,\n type ASRLanguage,\n type BaseEvent,\n type ClientEventType,\n type SessionConfig,\n type TurnDetectionConfig,\n type InputAudioTranscriptionConfig,\n type ClientEvent,\n type InputAudioBufferCommittedEvent,\n type ServerEventType,\n type ServerEvent,\n type SessionUpdateEvent,\n type InputAudioBufferAppendEvent,\n type InputAudioBufferCommitEvent,\n type SessionFinishEvent,\n type SessionCreatedEvent,\n type SessionUpdatedEvent,\n type SpeechStartedEvent,\n type SpeechStoppedEvent,\n type TranscriptionTextEvent,\n type TranscriptionCompletedEvent,\n type SessionFinishedEvent,\n type ErrorEvent,\n} from \"./asr-client\";\n\nexport type {\n ASRClient,\n ASRClientConfig,\n ASRLanguage,\n BaseEvent,\n ClientEventType,\n SessionConfig,\n TurnDetectionConfig,\n InputAudioTranscriptionConfig,\n ClientEvent,\n InputAudioBufferCommittedEvent,\n ServerEventType,\n ServerEvent,\n SessionUpdateEvent,\n InputAudioBufferAppendEvent,\n InputAudioBufferCommitEvent,\n SessionFinishEvent,\n SessionCreatedEvent,\n SessionUpdatedEvent,\n SpeechStartedEvent,\n SpeechStoppedEvent,\n TranscriptionTextEvent,\n TranscriptionCompletedEvent,\n SessionFinishedEvent,\n ErrorEvent,\n};\n\nexport { createASRClient };\n\nimport createASRHttpClient, {\n type ASRHttpClientConfig,\n type ASRHttpClient,\n type Recorder,\n type RecorderOptions,\n} from \"./http-asr-client\";\nexport type { ASRHttpClient, ASRHttpClientConfig, Recorder, RecorderOptions };\nexport { createASRHttpClient };\n","/**\n * ASR Realtime WebSocket Client for Qwen-ASR Realtime API\n *\n * WebSocket-based real-time speech recognition for streaming transcription.\n * Follows the Qwen-ASR Realtime API protocol with proper event handling.\n *\n * @example\n * ```typescript\n * const client = createASRClient({\n * language: \"zh\",\n * enableVAD: true,\n * onReady() {\n * console.log(\"ASR connected\");\n * },\n * onTranscript(text, isFinal) {\n * console.log(isFinal ? \"[Final]\" : \"[Interim]\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * await client.connect();\n * await client.startRecording();\n * // ... stop ...\n * await client.stopRecording();\n * await client.close();\n * ```\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\n\nexport type ASRLanguage =\n | \"zh\" // 中文（普通话、四川话、闽南语、吴语）\n | \"yue\" // 粤语\n | \"en\" // 英文\n | \"ja\" // 日语\n | \"de\" // 德语\n | \"ko\" // 韩语\n | \"ru\" // 俄语\n | \"fr\" // 法语\n | \"pt\" // 葡萄牙语\n | \"ar\" // 阿拉伯语\n | \"it\" // 意大利语\n | \"es\" // 西班牙语\n | \"hi\" // 印地语\n | \"id\" // 印尼语\n | \"th\" // 泰语\n | \"tr\" // 土耳其语\n | \"uk\" // 乌克兰语\n | \"vi\" // 越南语\n | \"cs\" // 捷克语\n | \"da\" // 丹麦语\n | \"fil\" // 菲律宾语\n | \"fi\" // 芬兰语\n | \"is\" // 冰岛语\n | \"ms\" // 马来语\n | \"no\" // 挪威语\n | \"pl\" // 波兰语\n | \"sv\"; // 瑞典语\n\n// Client event types (sent to server)\ntype ClientEventType =\n | \"session.update\"\n | \"input_audio_buffer.append\"\n | \"input_audio_buffer.commit\"\n | \"session.finish\";\n\n// Server event types (received from server)\ntype ServerEventType =\n | \"session.created\"\n | \"session.updated\"\n | \"input_audio_buffer.speech_started\"\n | \"input_audio_buffer.speech_stopped\"\n | \"input_audio_buffer.committed\"\n | \"conversation.item.input_audio_transcription.text\"\n | \"conversation.item.input_audio_transcription.completed\"\n | \"session.finished\"\n | \"error\";\n\n// Base event interface\ninterface BaseEvent {\n event_id: string;\n type: ClientEventType | ServerEventType;\n}\n\n// Client events\ninterface SessionUpdateEvent extends BaseEvent {\n type: \"session.update\";\n session: SessionConfig;\n}\n\ninterface InputAudioBufferAppendEvent extends BaseEvent {\n type: \"input_audio_buffer.append\";\n audio: string;\n}\n\ninterface InputAudioBufferCommitEvent extends BaseEvent {\n type: \"input_audio_buffer.commit\";\n}\n\ninterface SessionFinishEvent extends BaseEvent {\n type: \"session.finish\";\n}\n\ntype ClientEvent =\n | SessionUpdateEvent\n | InputAudioBufferAppendEvent\n | InputAudioBufferCommitEvent\n | SessionFinishEvent;\n\n// Server events\ninterface SessionCreatedEvent extends BaseEvent {\n type: \"session.created\";\n session: { id: string };\n}\n\ninterface SessionUpdatedEvent extends BaseEvent {\n type: \"session.updated\";\n session: SessionConfig;\n}\n\ninterface SpeechStartedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_started\";\n}\n\ninterface SpeechStoppedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_stopped\";\n}\n\ninterface InputAudioBufferCommittedEvent extends BaseEvent {\n type: \"input_audio_buffer.committed\";\n}\n\ninterface TranscriptionTextEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.text\";\n text?: string;\n stash?: string;\n transcript?: string;\n}\n\ninterface TranscriptionCompletedEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.completed\";\n text?: string;\n transcript?: string;\n}\n\ninterface SessionFinishedEvent extends BaseEvent {\n type: \"session.finished\";\n}\n\ninterface ErrorEvent extends BaseEvent {\n type: \"error\";\n error: {\n message: string;\n code?: string;\n };\n}\n\ntype ServerEvent =\n | SessionCreatedEvent\n | SessionUpdatedEvent\n | SpeechStartedEvent\n | SpeechStoppedEvent\n | InputAudioBufferCommittedEvent\n | TranscriptionTextEvent\n | TranscriptionCompletedEvent\n | SessionFinishedEvent\n | ErrorEvent;\n\n// Session configuration\ninterface TurnDetectionConfig {\n type: \"server_vad\";\n /** VAD检测阈值，推荐设为 0.0，默认值 0.2，范围 [-1, 1] */\n threshold?: number;\n /** VAD断句检测阈值(ms)，推荐设为 400，默认值 800，范围 [200, 6000] */\n silence_duration_ms?: number;\n}\n\ninterface InputAudioTranscriptionConfig {\n language?: ASRLanguage;\n}\n\ninterface SessionConfig {\n input_audio_format?: \"pcm\" | \"opus\";\n sample_rate?: 16000 | 8000;\n input_audio_transcription?: InputAudioTranscriptionConfig;\n turn_detection?: TurnDetectionConfig | null;\n}\n\ninterface RealtimeRecorder {\n start(onAudio: (base64: string) => void): Promise<void>;\n stop(): Promise<void>;\n}\n\nasync function createRealtimeRecorder(): Promise<RealtimeRecorder> {\n let stream: MediaStream | null = null;\n let ctx: AudioContext | null = null;\n let source: MediaStreamAudioSourceNode | null = null;\n let processor: ScriptProcessorNode | null = null;\n\n if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {\n throw new Error(\"未找到麦克风或无权限\");\n }\n\n return {\n async start(onAudio) {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: { sampleRate: 16000, channelCount: 1, echoCancellation: true },\n });\n\n log(\"✅ 麦克风已启动\", \"success\");\n log(\"💬 请对着麦克风说话，实时识别中...\", \"success\");\n\n ctx = new AudioContext({ sampleRate: 16000 });\n source = ctx.createMediaStreamSource(stream);\n processor = ctx.createScriptProcessor(4096, 1, 1);\n\n processor.onaudioprocess = (e) => {\n const inputData = e.inputBuffer.getChannelData(0);\n const pcm = new Int16Array(inputData.length);\n for (let i = 0; i < inputData.length; i++) {\n const s = Math.max(-1, Math.min(1, inputData[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n\n const bytes = new Uint8Array(pcm.buffer);\n let binary = \"\";\n for (let i = 0; i < bytes.length; i++) {\n binary += String.fromCharCode(bytes[i] || 0);\n }\n\n onAudio(btoa(binary));\n };\n\n source.connect(processor);\n processor.connect(ctx.destination);\n },\n\n async stop() {\n stream?.getTracks().forEach((t) => t.stop());\n source?.disconnect();\n processor?.disconnect();\n if (ctx) {\n await ctx.close();\n }\n stream = null;\n ctx = null;\n source = null;\n processor = null;\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\n// Generate unique event ID\nlet eventIdCounter = 0;\nfunction generateEventId(): string {\n return `event_${Date.now()}_${++eventIdCounter}`;\n}\n\nexport interface ASRClientConfig {\n /**\n * Audio format\n * @default \"pcm\"\n */\n audioFormat?: \"pcm\" | \"opus\";\n /**\n * Sample rate in Hz\n * @default 16000\n * @description 支持 16000 和 8000。设置为 8000 时，服务端会先升采样到16000Hz再进行识别，可能引入微小延迟。\n */\n sampleRate?: 16000 | 8000;\n /**\n * Audio source language\n * @default \"zh\"\n * @description 支持多种语言，包括 zh（中文）、yue（粤语）、en（英文）、ja（日语）等\n */\n language?: ASRLanguage;\n /**\n * Enable VAD (Voice Activity Detection) mode\n * @default true\n * @description true = VAD模式（服务端自动检测语音开始/结束），false = Manual模式（客户端手动控制）\n */\n enableVAD?: boolean;\n /**\n * VAD detection threshold\n * @default 0.2\n * @description 推荐设为 0.0。取值范围 [-1, 1]。较低的阈值会提高 VAD 的灵敏度。\n */\n vadThreshold?: number;\n /**\n * VAD silence duration threshold in milliseconds\n * @default 800\n * @description 推荐设为 400。取值范围 [200, 6000]。静音持续时长超过该阈值将被认为是语句结束。\n */\n vadSilenceDurationMs?: number;\n /**\n * Get access token for WebSocket authentication\n */\n getAccessToken?: () => string | null;\n /**\n * Called when connection is ready (session.created received and session.update sent)\n */\n onReady?: () => void;\n /**\n * Called when speech is detected (VAD mode only)\n */\n onSpeechStart?: () => void;\n /**\n * Called when speech stops (VAD mode only)\n */\n onSpeechEnd?: () => void;\n /**\n * Called on transcript result\n * @param text - Transcribed text\n * @param isFinal - Whether this is the final result\n */\n onTranscript?: (text: string, isFinal: boolean) => void;\n /**\n * Called when audio buffer is committed (non-VAD mode only)\n */\n onAudioBufferCommitted?: () => void;\n /**\n * Called when session is finished\n */\n onSessionFinished?: () => void;\n /**\n * Called on error\n */\n onError?: (error: Error) => void;\n /**\n * Called on close\n */\n onClose?: () => void;\n}\n\nexport interface ASRClient {\n /** Connect to ASR service and establish session */\n connect(): Promise<void>;\n /** Start recording from microphone */\n startRecording(): Promise<void>;\n /**\n * Stop recording\n * @description In non-VAD mode, this triggers recognition by sending input_audio_buffer.commit\n */\n stopRecording(): Promise<void>;\n /**\n * Close connection gracefully\n * @description Sends session.finish and waits for session.finished before closing\n */\n close(): Promise<void>;\n /**\n * Check if currently recording\n */\n isRecording(): boolean;\n /**\n * Check if connected to server\n */\n isConnected(): boolean;\n}\n\nfunction createASRClient(config: ASRClientConfig): ASRClient {\n const {\n onReady,\n onSpeechStart,\n onSpeechEnd,\n onTranscript,\n onAudioBufferCommitted,\n onSessionFinished,\n onError,\n onClose,\n getAccessToken,\n audioFormat = \"pcm\",\n sampleRate = 16000,\n language = \"zh\",\n enableVAD = true,\n vadThreshold = 0.2,\n vadSilenceDurationMs = 400,\n } = config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\n let isRecordingFlag = false;\n let isClosing = false;\n\n const path = ASR_PATH;\n\n /**\n * Send event to server\n */\n function sendEvent(event: ClientEvent): void {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n ws.send(JSON.stringify(event));\n }\n\n /**\n * Build session configuration\n */\n function buildSessionConfig(): SessionConfig {\n const sessionConfig: SessionConfig = {\n input_audio_format: audioFormat,\n sample_rate: sampleRate,\n input_audio_transcription: {\n language,\n },\n };\n\n if (enableVAD) {\n sessionConfig.turn_detection = {\n type: \"server_vad\",\n threshold: vadThreshold,\n silence_duration_ms: vadSilenceDurationMs,\n };\n } else {\n // Non-VAD mode: set turn_detection to null\n sessionConfig.turn_detection = null;\n }\n\n return sessionConfig;\n }\n\n /**\n * Send session.update event\n */\n function sendSessionUpdate(): void {\n const event: SessionUpdateEvent = {\n event_id: generateEventId(),\n type: \"session.update\",\n session: buildSessionConfig(),\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.append event\n */\n function sendAudioBufferAppend(audio: string): void {\n const event: InputAudioBufferAppendEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.append\",\n audio,\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.commit event (for non-VAD mode)\n */\n function sendAudioBufferCommit(): void {\n const event: InputAudioBufferCommitEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.commit\",\n };\n sendEvent(event);\n }\n\n /**\n * Send session.finish event\n */\n function sendSessionFinish(): void {\n const event: SessionFinishEvent = {\n event_id: generateEventId(),\n type: \"session.finish\",\n };\n sendEvent(event);\n }\n\n /**\n * Handle server events\n */\n function handleServerEvent(data: ServerEvent): void {\n switch (data.type) {\n case \"session.created\":\n // Send session.update immediately after session.created\n try {\n sendSessionUpdate();\n } catch (err) {\n onError?.(\n new Error(\n \"Failed to send session.update: \" + (err instanceof Error ? err.message : String(err))\n )\n );\n }\n break;\n\n case \"session.updated\":\n // Server confirmed session configuration\n onReady?.();\n break;\n\n case \"input_audio_buffer.speech_started\":\n onSpeechStart?.();\n break;\n\n case \"input_audio_buffer.speech_stopped\":\n onSpeechEnd?.();\n break;\n\n case \"input_audio_buffer.committed\":\n onAudioBufferCommitted?.();\n break;\n\n case \"conversation.item.input_audio_transcription.text\":\n onTranscript?.(data.text || data.stash || data.transcript || \"\", false);\n break;\n\n case \"conversation.item.input_audio_transcription.completed\":\n onTranscript?.(data.text || data.transcript || \"\", true);\n break;\n\n case \"session.finished\":\n onSessionFinished?.();\n // Close WebSocket after receiving session.finished\n close();\n break;\n\n case \"error\":\n const err = new Error(data.error?.message || \"ASR error\");\n onError?.(err);\n break;\n\n default:\n // Unknown event type\n console.warn(\"[ASR] Unknown server event:\", (data as ServerEvent).type);\n }\n }\n\n async function connect(): Promise<void> {\n // Build WebSocket URL with optional token parameter\n let wsUrl = path;\n if (getAccessToken) {\n const token = getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n wsUrl = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n // Support both ws:// and wss:// protocols\n if (typeof window !== \"undefined\" && window.location) {\n const protocol = window.location.protocol === \"https:\" ? \"wss:\" : \"ws:\";\n if (!wsUrl.startsWith(\"ws://\") && !wsUrl.startsWith(\"wss://\")) {\n wsUrl = `${protocol}//${window.location.host}${wsUrl}`;\n }\n }\n\n ws = new WebSocket(wsUrl);\n\n return new Promise((resolve, reject) => {\n if (!ws) {\n reject(new Error(\"Failed to create WebSocket\"));\n return;\n }\n\n ws.onopen = () => {\n log(\"WebSocket connected\", \"success\");\n };\n\n ws.onmessage = (event) => {\n try {\n const data = JSON.parse(event.data) as ServerEvent;\n handleServerEvent(data);\n\n // Resolve on session.updated (ready state)\n if (data.type === \"session.updated\") {\n resolve();\n }\n } catch (err) {\n const error = new Error(\n \"Failed to parse server message: \" + (err instanceof Error ? err.message : String(err))\n );\n onError?.(error);\n reject(error);\n }\n };\n\n ws.onerror = (error) => {\n console.error(\"WebSocket error:\", error);\n const err = new Error(\"WebSocket error\");\n onError?.(err);\n reject(err);\n };\n\n ws.onclose = () => {\n isRecordingFlag = false;\n ws = null;\n onClose?.();\n };\n });\n }\n\n async function startRecording(): Promise<void> {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n\n if (isRecordingFlag) {\n return;\n }\n\n try {\n recorder = await createRealtimeRecorder();\n isRecordingFlag = true;\n\n await recorder.start((audio) => {\n if (!ws || ws.readyState !== WebSocket.OPEN) return;\n try {\n sendAudioBufferAppend(audio);\n } catch (err) {\n console.error(\"[ASR] Failed to send audio:\", err);\n }\n });\n } catch (err) {\n console.error(\"[ASR] Failed to start recorder:\", err);\n onError?.(err instanceof Error ? err : new Error(String(err)));\n throw err;\n }\n }\n\n async function stopRecording(): Promise<void> {\n if (!isRecordingFlag) {\n return;\n }\n\n try {\n await recorder?.stop();\n } catch (err) {\n console.error(\"[ASR] Error stopping recorder:\", err);\n }\n recorder = null;\n isRecordingFlag = false;\n\n // In non-VAD mode, send commit to trigger recognition\n if (!enableVAD && ws?.readyState === WebSocket.OPEN) {\n try {\n sendAudioBufferCommit();\n } catch (err) {\n console.error(\"[ASR] Failed to send commit:\", err);\n }\n }\n }\n\n async function close(): Promise<void> {\n if (isClosing) {\n return;\n }\n\n isClosing = true;\n\n // Stop recording first\n await stopRecording();\n\n // Send session.finish if connected\n if (ws?.readyState === WebSocket.OPEN) {\n try {\n sendSessionFinish();\n // Wait a bit for session.finished response\n await new Promise((resolve) => setTimeout(resolve, 1000));\n } catch (err) {\n console.error(\"[ASR] Failed to send session.finish:\", err);\n }\n }\n\n // Close WebSocket\n if (ws && ws?.readyState !== WebSocket.CLOSING && ws?.readyState !== WebSocket.CLOSED) {\n ws?.close();\n }\n ws = null;\n isClosing = false;\n }\n\n function isRecording(): boolean {\n return isRecordingFlag;\n }\n\n function isConnected(): boolean {\n return ws !== null && ws.readyState === WebSocket.OPEN;\n }\n\n return {\n connect,\n startRecording,\n stopRecording,\n close,\n isRecording,\n isConnected,\n };\n}\n\nexport default (authConfig: Pick<ASRClientConfig, \"getAccessToken\">) => (config: ASRClientConfig) =>\n createASRClient({ ...authConfig, ...config });\n\nexport type {\n BaseEvent,\n ClientEventType,\n SessionConfig,\n TurnDetectionConfig,\n InputAudioTranscriptionConfig,\n ClientEvent,\n InputAudioBufferCommittedEvent,\n ServerEventType,\n ServerEvent,\n SessionUpdateEvent,\n InputAudioBufferAppendEvent,\n InputAudioBufferCommitEvent,\n SessionFinishEvent,\n SessionCreatedEvent,\n SessionUpdatedEvent,\n SpeechStartedEvent,\n SpeechStoppedEvent,\n TranscriptionTextEvent,\n TranscriptionCompletedEvent,\n SessionFinishedEvent,\n ErrorEvent,\n};\n","/**\n * HTTP ASR Client - Press-to-talk style speech recognition\n *\n * HTTP-based speech recognition suitable for press-to-talk scenarios where you hold to speak\n * and release to recognize. Good for voice messages, voice search, etc.\n *\n * @example\n * ```typescript\n * const client = createASRHttpClient({\n * onRecordingStart() {\n * console.log(\"Recording started\");\n * },\n * onRecordingStop() {\n * console.log(\"Recording stopped\");\n * },\n * onResult(text) {\n * console.log(\"Recognized:\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * // Hold to speak, release to recognize\n * await client.startRecording();\n * // ... stop ...\n * const result = await client.stopRecording();\n * ```\n */\n\nimport { createHttpClient, type HttpClient } from \"@amaster.ai/http-client\";\n\nconst ASR_HTTP_PATH =\n \"/api/proxy/builtin/platform/qwen-asr/compatible-mode/v1/chat/completions\";\n\n// 录音内核\nconst RECORDER_WORKLET = `\nclass RecorderProcessor extends AudioWorkletProcessor {\n process(inputs) {\n const input = inputs[0];\n if (input && input[0]) {\n this.port.postMessage(input[0].slice(0));\n }\n return true;\n }\n}\nregisterProcessor('recorder-processor', RecorderProcessor);\n`;\n\ntype Status = \"idle\" | \"recording\" | \"recognizing\";\n\nexport interface Recorder {\n /** Start recording */\n start(): Promise<void>;\n /**\n * Stop recording and get base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @returns Base64-encoded WAV audio data\n */\n stop(): Promise<void>;\n}\n\nexport interface RecorderOptions {\n /** Called when recording starts */\n onStart?: () => void;\n /**\n * Called when recording stops, with base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @param base64 - Base64-encoded WAV audio data\n * @returns void\n */\n onStop?: (base64: string) => void;\n onError?: (error: Error) => void;\n}\n\n// 创建录音器\nasync function createWebRecorder(props?: RecorderOptions): Promise<Recorder> {\n let stream: MediaStream;\n let ctx: AudioContext;\n let node: AudioWorkletNode;\n let source: MediaStreamAudioSourceNode;\n const chunks: Int16Array[] = [];\n\n const cleanup = () => {\n try {\n source?.disconnect();\n node?.disconnect();\n stream?.getTracks().forEach((t) => t.stop());\n ctx?.close();\n } catch (e) {}\n };\n\n return {\n async start() {\n try {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: {\n channelCount: 1,\n echoCancellation: true,\n noiseSuppression: true,\n autoGainControl: true,\n },\n });\n\n ctx = new AudioContext();\n\n const blob = new Blob([RECORDER_WORKLET], {\n type: \"application/javascript\",\n });\n const url = URL.createObjectURL(blob);\n await ctx.audioWorklet.addModule(url);\n URL.revokeObjectURL(url);\n\n source = ctx.createMediaStreamSource(stream);\n node = new AudioWorkletNode(ctx, \"recorder-processor\");\n\n node.port.onmessage = (e) => {\n const input = e.data as Float32Array;\n const pcm = new Int16Array(input.length);\n for (let i = 0; i < input.length; i++) {\n const s = Math.max(-1, Math.min(1, input[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n chunks.push(pcm);\n };\n\n source.connect(node);\n\n props?.onStart?.();\n } catch (error) {\n props?.onError?.(\n error instanceof Error ? error : new Error(String(error)),\n );\n cleanup();\n }\n },\n\n async stop() {\n cleanup();\n\n const total = chunks.reduce((s, c) => s + c.length, 0);\n const pcm = new Int16Array(total);\n let offset = 0;\n for (const c of chunks) {\n pcm.set(c, offset);\n offset += c.length;\n }\n\n const result = { pcm, sampleRate: ctx?.sampleRate ?? 16000 };\n const base64 = await blobToBase64(\n pcmToWav(result.pcm, 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const response = await this.http.request({\n url: this.path,\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n data: JSON.stringify({\n model: \"qwen3-asr-flash\",\n messages: [\n {\n role: \"user\",\n content: [\n {\n type: \"input_audio\",\n input_audio: { data: `data:audio/wav;base64,${base64}` },\n },\n ],\n },\n ],\n }),\n });\n\n return (response as any)?.data?.choices?.[0]?.message?.content || \"\";\n } catch (e) {\n console.error(\"ASR recognition error:\", e);\n return \"\";\n } finally {\n this.recognizing = false;\n }\n }\n\n async recordAndRecognize(ms: number) {\n await this.startRecording();\n await new Promise((r) => setTimeout(r, ms));\n await this.stopRecording();\n }\n\n async recognizeUrl(url: string) {\n try {\n const res = await this.http.request({\n url: this.path,\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n data: JSON.stringify({\n model: \"qwen3-asr-flash\",\n messages: [\n {\n role: \"user\",\n 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package/dist/index.d.cts CHANGED
@@ -297,4 +297,4 @@ interface ASRHttpClient {
297
297
  }
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298
  declare const _default: (authConfig: Pick<ASRHttpClientConfig, "getAccessToken" | "http">) => (config: ASRHttpClientConfig) => ASRHttpClient;
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- export { type ASRClient, type ASRClientConfig, type ASRHttpClient, type ASRHttpClientConfig, type ASRLanguage, type ClientEvent, type ErrorEvent, type InputAudioBufferAppendEvent, type InputAudioBufferCommitEvent, type InputAudioTranscriptionConfig, type ServerEvent, type SessionConfig, type SessionCreatedEvent, type SessionFinishEvent, type SessionFinishedEvent, type SessionUpdateEvent, type SessionUpdatedEvent, type TranscriptionCompletedEvent, type TranscriptionTextEvent, type TurnDetectionConfig, _default$1 as createASRClient, _default as createASRHttpClient };
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+ export { type ASRClient, type ASRClientConfig, type ASRHttpClient, type ASRHttpClientConfig, type ASRLanguage, type BaseEvent, type ClientEvent, type ClientEventType, type ErrorEvent, type InputAudioBufferAppendEvent, type InputAudioBufferCommitEvent, type InputAudioBufferCommittedEvent, type InputAudioTranscriptionConfig, type Recorder, type RecorderOptions, type ServerEvent, type ServerEventType, type SessionConfig, type SessionCreatedEvent, type SessionFinishEvent, type SessionFinishedEvent, type SessionUpdateEvent, type SessionUpdatedEvent, type SpeechStartedEvent, type SpeechStoppedEvent, type TranscriptionCompletedEvent, type TranscriptionTextEvent, type TurnDetectionConfig, _default$1 as createASRClient, _default as createASRHttpClient };
package/dist/index.d.ts CHANGED
@@ -297,4 +297,4 @@ interface ASRHttpClient {
297
297
  }
298
298
  declare const _default: (authConfig: Pick<ASRHttpClientConfig, "getAccessToken" | "http">) => (config: ASRHttpClientConfig) => ASRHttpClient;
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- export { type ASRClient, type ASRClientConfig, type ASRHttpClient, type ASRHttpClientConfig, type ASRLanguage, type ClientEvent, type ErrorEvent, type InputAudioBufferAppendEvent, type InputAudioBufferCommitEvent, type InputAudioTranscriptionConfig, type ServerEvent, type SessionConfig, type SessionCreatedEvent, type SessionFinishEvent, type SessionFinishedEvent, type SessionUpdateEvent, type SessionUpdatedEvent, type TranscriptionCompletedEvent, type TranscriptionTextEvent, type TurnDetectionConfig, _default$1 as createASRClient, _default as createASRHttpClient };
300
+ export { type ASRClient, type ASRClientConfig, type ASRHttpClient, type ASRHttpClientConfig, type ASRLanguage, type BaseEvent, type ClientEvent, type ClientEventType, type ErrorEvent, type InputAudioBufferAppendEvent, type InputAudioBufferCommitEvent, type InputAudioBufferCommittedEvent, type InputAudioTranscriptionConfig, type Recorder, type RecorderOptions, type ServerEvent, type ServerEventType, type SessionConfig, type SessionCreatedEvent, type SessionFinishEvent, type SessionFinishedEvent, type SessionUpdateEvent, type SessionUpdatedEvent, type SpeechStartedEvent, type SpeechStoppedEvent, type TranscriptionCompletedEvent, type TranscriptionTextEvent, type TurnDetectionConfig, _default$1 as createASRClient, _default as createASRHttpClient };
package/dist/index.js.map CHANGED
@@ -1 +1 @@
1
- {"version":3,"sources":["../src/asr-client.ts","../src/http-asr-client.ts"],"sourcesContent":["/**\n * ASR Realtime WebSocket Client for Qwen-ASR Realtime API\n *\n * WebSocket-based real-time speech recognition for streaming transcription.\n * Follows the Qwen-ASR Realtime API protocol with proper event handling.\n *\n * @example\n * ```typescript\n * const client = createASRClient({\n * language: \"zh\",\n * enableVAD: true,\n * onReady() {\n * console.log(\"ASR connected\");\n * },\n * onTranscript(text, isFinal) {\n * console.log(isFinal ? \"[Final]\" : \"[Interim]\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * await client.connect();\n * await client.startRecording();\n * // ... stop ...\n * await client.stopRecording();\n * await client.close();\n * ```\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\n\nexport type ASRLanguage =\n | \"zh\" // 中文（普通话、四川话、闽南语、吴语）\n | \"yue\" // 粤语\n | \"en\" // 英文\n | \"ja\" // 日语\n | \"de\" // 德语\n | \"ko\" // 韩语\n | \"ru\" // 俄语\n | \"fr\" // 法语\n | \"pt\" // 葡萄牙语\n | \"ar\" // 阿拉伯语\n | \"it\" // 意大利语\n | \"es\" // 西班牙语\n | \"hi\" // 印地语\n | \"id\" // 印尼语\n | \"th\" // 泰语\n | \"tr\" // 土耳其语\n | \"uk\" // 乌克兰语\n | \"vi\" // 越南语\n | \"cs\" // 捷克语\n | \"da\" // 丹麦语\n | \"fil\" // 菲律宾语\n | \"fi\" // 芬兰语\n | \"is\" // 冰岛语\n | \"ms\" // 马来语\n | \"no\" // 挪威语\n | \"pl\" // 波兰语\n | \"sv\"; // 瑞典语\n\n// Client event types (sent to server)\ntype ClientEventType =\n | \"session.update\"\n | \"input_audio_buffer.append\"\n | \"input_audio_buffer.commit\"\n | \"session.finish\";\n\n// Server event types (received from server)\ntype ServerEventType =\n | \"session.created\"\n | \"session.updated\"\n | \"input_audio_buffer.speech_started\"\n | \"input_audio_buffer.speech_stopped\"\n | \"input_audio_buffer.committed\"\n | \"conversation.item.input_audio_transcription.text\"\n | \"conversation.item.input_audio_transcription.completed\"\n | \"session.finished\"\n | \"error\";\n\n// Base event interface\ninterface BaseEvent {\n event_id: string;\n type: ClientEventType | ServerEventType;\n}\n\n// Client events\ninterface SessionUpdateEvent extends BaseEvent {\n type: \"session.update\";\n session: SessionConfig;\n}\n\ninterface InputAudioBufferAppendEvent extends BaseEvent {\n type: \"input_audio_buffer.append\";\n audio: string;\n}\n\ninterface InputAudioBufferCommitEvent extends BaseEvent {\n type: \"input_audio_buffer.commit\";\n}\n\ninterface SessionFinishEvent extends BaseEvent {\n type: \"session.finish\";\n}\n\ntype ClientEvent =\n | SessionUpdateEvent\n | InputAudioBufferAppendEvent\n | InputAudioBufferCommitEvent\n | SessionFinishEvent;\n\n// Server events\ninterface SessionCreatedEvent extends BaseEvent {\n type: \"session.created\";\n session: { id: string };\n}\n\ninterface SessionUpdatedEvent extends BaseEvent {\n type: \"session.updated\";\n session: SessionConfig;\n}\n\ninterface SpeechStartedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_started\";\n}\n\ninterface SpeechStoppedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_stopped\";\n}\n\ninterface InputAudioBufferCommittedEvent extends BaseEvent {\n type: \"input_audio_buffer.committed\";\n}\n\ninterface TranscriptionTextEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.text\";\n text?: string;\n stash?: string;\n transcript?: string;\n}\n\ninterface TranscriptionCompletedEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.completed\";\n text?: string;\n transcript?: string;\n}\n\ninterface SessionFinishedEvent extends BaseEvent {\n type: \"session.finished\";\n}\n\ninterface ErrorEvent extends BaseEvent {\n type: \"error\";\n error: {\n message: string;\n code?: string;\n };\n}\n\ntype ServerEvent =\n | SessionCreatedEvent\n | SessionUpdatedEvent\n | SpeechStartedEvent\n | SpeechStoppedEvent\n | InputAudioBufferCommittedEvent\n | TranscriptionTextEvent\n | TranscriptionCompletedEvent\n | SessionFinishedEvent\n | ErrorEvent;\n\n// Session configuration\ninterface TurnDetectionConfig {\n type: \"server_vad\";\n /** VAD检测阈值，推荐设为 0.0，默认值 0.2，范围 [-1, 1] */\n threshold?: number;\n /** VAD断句检测阈值(ms)，推荐设为 400，默认值 800，范围 [200, 6000] */\n silence_duration_ms?: number;\n}\n\ninterface InputAudioTranscriptionConfig {\n language?: ASRLanguage;\n}\n\ninterface SessionConfig {\n input_audio_format?: \"pcm\" | \"opus\";\n sample_rate?: 16000 | 8000;\n input_audio_transcription?: InputAudioTranscriptionConfig;\n turn_detection?: TurnDetectionConfig | null;\n}\n\ninterface RealtimeRecorder {\n start(onAudio: (base64: string) => void): Promise<void>;\n stop(): Promise<void>;\n}\n\nasync function createRealtimeRecorder(): Promise<RealtimeRecorder> {\n let stream: MediaStream | null = null;\n let ctx: AudioContext | null = null;\n let source: MediaStreamAudioSourceNode | null = null;\n let processor: ScriptProcessorNode | null = null;\n\n if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {\n throw new Error(\"未找到麦克风或无权限\");\n }\n\n return {\n async start(onAudio) {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: { sampleRate: 16000, channelCount: 1, echoCancellation: true },\n });\n\n log(\"✅ 麦克风已启动\", \"success\");\n log(\"💬 请对着麦克风说话，实时识别中...\", \"success\");\n\n ctx = new AudioContext({ sampleRate: 16000 });\n source = ctx.createMediaStreamSource(stream);\n processor = ctx.createScriptProcessor(4096, 1, 1);\n\n processor.onaudioprocess = (e) => {\n const inputData = e.inputBuffer.getChannelData(0);\n const pcm = new Int16Array(inputData.length);\n for (let i = 0; i < inputData.length; i++) {\n const s = Math.max(-1, Math.min(1, inputData[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n\n const bytes = new Uint8Array(pcm.buffer);\n let binary = \"\";\n for (let i = 0; i < bytes.length; i++) {\n binary += String.fromCharCode(bytes[i] || 0);\n }\n\n onAudio(btoa(binary));\n };\n\n source.connect(processor);\n processor.connect(ctx.destination);\n },\n\n async stop() {\n stream?.getTracks().forEach((t) => t.stop());\n source?.disconnect();\n processor?.disconnect();\n if (ctx) {\n await ctx.close();\n }\n stream = null;\n ctx = null;\n source = null;\n processor = null;\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\n// Generate unique event ID\nlet eventIdCounter = 0;\nfunction generateEventId(): string {\n return `event_${Date.now()}_${++eventIdCounter}`;\n}\n\nexport interface ASRClientConfig {\n /**\n * Audio format\n * @default \"pcm\"\n */\n audioFormat?: \"pcm\" | \"opus\";\n /**\n * Sample rate in Hz\n * @default 16000\n * @description 支持 16000 和 8000。设置为 8000 时，服务端会先升采样到16000Hz再进行识别，可能引入微小延迟。\n */\n sampleRate?: 16000 | 8000;\n /**\n * Audio source language\n * @default \"zh\"\n * @description 支持多种语言，包括 zh（中文）、yue（粤语）、en（英文）、ja（日语）等\n */\n language?: ASRLanguage;\n /**\n * Enable VAD (Voice Activity Detection) mode\n * @default true\n * @description true = VAD模式（服务端自动检测语音开始/结束），false = Manual模式（客户端手动控制）\n */\n enableVAD?: boolean;\n /**\n * VAD detection threshold\n * @default 0.2\n * @description 推荐设为 0.0。取值范围 [-1, 1]。较低的阈值会提高 VAD 的灵敏度。\n */\n vadThreshold?: number;\n /**\n * VAD silence duration threshold in milliseconds\n * @default 800\n * @description 推荐设为 400。取值范围 [200, 6000]。静音持续时长超过该阈值将被认为是语句结束。\n */\n vadSilenceDurationMs?: number;\n /**\n * Get access token for WebSocket authentication\n */\n getAccessToken?: () => string | null;\n /**\n * Called when connection is ready (session.created received and session.update sent)\n */\n onReady?: () => void;\n /**\n * Called when speech is detected (VAD mode only)\n */\n onSpeechStart?: () => void;\n /**\n * Called when speech stops (VAD mode only)\n */\n onSpeechEnd?: () => void;\n /**\n * Called on transcript result\n * @param text - Transcribed text\n * @param isFinal - Whether this is the final result\n */\n onTranscript?: (text: string, isFinal: boolean) => void;\n /**\n * Called when audio buffer is committed (non-VAD mode only)\n */\n onAudioBufferCommitted?: () => void;\n /**\n * Called when session is finished\n */\n onSessionFinished?: () => void;\n /**\n * Called on error\n */\n onError?: (error: Error) => void;\n /**\n * Called on close\n */\n onClose?: () => void;\n}\n\nexport interface ASRClient {\n /** Connect to ASR service and establish session */\n connect(): Promise<void>;\n /** Start recording from microphone */\n startRecording(): Promise<void>;\n /**\n * Stop recording\n * @description In non-VAD mode, this triggers recognition by sending input_audio_buffer.commit\n */\n stopRecording(): Promise<void>;\n /**\n * Close connection gracefully\n * @description Sends session.finish and waits for session.finished before closing\n */\n close(): Promise<void>;\n /**\n * Check if currently recording\n */\n isRecording(): boolean;\n /**\n * Check if connected to server\n */\n isConnected(): boolean;\n}\n\nfunction createASRClient(config: ASRClientConfig): ASRClient {\n const {\n onReady,\n onSpeechStart,\n onSpeechEnd,\n onTranscript,\n onAudioBufferCommitted,\n onSessionFinished,\n onError,\n onClose,\n getAccessToken,\n audioFormat = \"pcm\",\n sampleRate = 16000,\n language = \"zh\",\n enableVAD = true,\n vadThreshold = 0.2,\n vadSilenceDurationMs = 400,\n } = config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\n let isRecordingFlag = false;\n let isClosing = false;\n\n const path = ASR_PATH;\n\n /**\n * Send event to server\n */\n function sendEvent(event: ClientEvent): void {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n ws.send(JSON.stringify(event));\n }\n\n /**\n * Build session configuration\n */\n function buildSessionConfig(): SessionConfig {\n const sessionConfig: SessionConfig = {\n input_audio_format: audioFormat,\n sample_rate: sampleRate,\n input_audio_transcription: {\n language,\n },\n };\n\n if (enableVAD) {\n sessionConfig.turn_detection = {\n type: \"server_vad\",\n threshold: vadThreshold,\n silence_duration_ms: vadSilenceDurationMs,\n };\n } else {\n // Non-VAD mode: set turn_detection to null\n sessionConfig.turn_detection = null;\n }\n\n return sessionConfig;\n }\n\n /**\n * Send session.update event\n */\n function sendSessionUpdate(): void {\n const event: SessionUpdateEvent = {\n event_id: generateEventId(),\n type: \"session.update\",\n session: buildSessionConfig(),\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.append event\n */\n function sendAudioBufferAppend(audio: string): void {\n const event: InputAudioBufferAppendEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.append\",\n audio,\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.commit event (for non-VAD mode)\n */\n function sendAudioBufferCommit(): void {\n const event: InputAudioBufferCommitEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.commit\",\n };\n sendEvent(event);\n }\n\n /**\n * Send session.finish event\n */\n function sendSessionFinish(): void {\n const event: SessionFinishEvent = {\n event_id: generateEventId(),\n type: \"session.finish\",\n };\n sendEvent(event);\n }\n\n /**\n * Handle server events\n */\n function handleServerEvent(data: ServerEvent): void {\n switch (data.type) {\n case \"session.created\":\n // Send session.update immediately after session.created\n try {\n sendSessionUpdate();\n } catch (err) {\n onError?.(\n new Error(\n \"Failed to send session.update: \" + (err instanceof Error ? err.message : String(err))\n )\n );\n }\n break;\n\n case \"session.updated\":\n // Server confirmed session configuration\n onReady?.();\n break;\n\n case \"input_audio_buffer.speech_started\":\n onSpeechStart?.();\n break;\n\n case \"input_audio_buffer.speech_stopped\":\n onSpeechEnd?.();\n break;\n\n case \"input_audio_buffer.committed\":\n onAudioBufferCommitted?.();\n break;\n\n case \"conversation.item.input_audio_transcription.text\":\n onTranscript?.(data.text || data.stash || data.transcript || \"\", false);\n break;\n\n case \"conversation.item.input_audio_transcription.completed\":\n onTranscript?.(data.text || data.transcript || \"\", true);\n break;\n\n case \"session.finished\":\n onSessionFinished?.();\n // Close WebSocket after receiving session.finished\n close();\n break;\n\n case \"error\":\n const err = new Error(data.error?.message || \"ASR error\");\n onError?.(err);\n break;\n\n default:\n // Unknown event type\n console.warn(\"[ASR] Unknown server event:\", (data as ServerEvent).type);\n }\n }\n\n async function connect(): Promise<void> {\n // Build WebSocket URL with optional token parameter\n let wsUrl = path;\n if (getAccessToken) {\n const token = getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n wsUrl = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n // Support both ws:// and wss:// protocols\n if (typeof window !== \"undefined\" && window.location) {\n const protocol = window.location.protocol === \"https:\" ? \"wss:\" : \"ws:\";\n if (!wsUrl.startsWith(\"ws://\") && !wsUrl.startsWith(\"wss://\")) {\n wsUrl = `${protocol}//${window.location.host}${wsUrl}`;\n }\n }\n\n ws = new WebSocket(wsUrl);\n\n return new Promise((resolve, reject) => {\n if (!ws) {\n reject(new Error(\"Failed to create WebSocket\"));\n return;\n }\n\n ws.onopen = () => {\n log(\"WebSocket connected\", \"success\");\n };\n\n ws.onmessage = (event) => {\n try {\n const data = JSON.parse(event.data) as ServerEvent;\n handleServerEvent(data);\n\n // Resolve on session.updated (ready state)\n if (data.type === \"session.updated\") {\n resolve();\n }\n } catch (err) {\n const error = new Error(\n \"Failed to parse server message: \" + (err instanceof Error ? err.message : String(err))\n );\n onError?.(error);\n reject(error);\n }\n };\n\n ws.onerror = (error) => {\n console.error(\"WebSocket error:\", error);\n const err = new Error(\"WebSocket error\");\n onError?.(err);\n reject(err);\n };\n\n ws.onclose = () => {\n isRecordingFlag = false;\n ws = null;\n onClose?.();\n };\n });\n }\n\n async function startRecording(): Promise<void> {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n\n if (isRecordingFlag) {\n return;\n }\n\n try {\n recorder = await createRealtimeRecorder();\n isRecordingFlag = true;\n\n await recorder.start((audio) => {\n if (!ws || ws.readyState !== WebSocket.OPEN) return;\n try {\n sendAudioBufferAppend(audio);\n } catch (err) {\n console.error(\"[ASR] Failed to send audio:\", err);\n }\n });\n } catch (err) {\n console.error(\"[ASR] Failed to start recorder:\", err);\n onError?.(err instanceof Error ? err : new Error(String(err)));\n throw err;\n }\n }\n\n async function stopRecording(): Promise<void> {\n if (!isRecordingFlag) {\n return;\n }\n\n try {\n await recorder?.stop();\n } catch (err) {\n console.error(\"[ASR] Error stopping recorder:\", err);\n }\n recorder = null;\n isRecordingFlag = false;\n\n // In non-VAD mode, send commit to trigger recognition\n if (!enableVAD && ws?.readyState === WebSocket.OPEN) {\n try {\n sendAudioBufferCommit();\n } catch (err) {\n console.error(\"[ASR] Failed to send commit:\", err);\n }\n }\n }\n\n async function close(): Promise<void> {\n if (isClosing) {\n return;\n }\n\n isClosing = true;\n\n // Stop recording first\n await stopRecording();\n\n // Send session.finish if connected\n if (ws?.readyState === WebSocket.OPEN) {\n try {\n sendSessionFinish();\n // Wait a bit for session.finished response\n await new Promise((resolve) => setTimeout(resolve, 1000));\n } catch (err) {\n console.error(\"[ASR] Failed to send session.finish:\", err);\n }\n }\n\n // Close WebSocket\n if (ws && ws?.readyState !== WebSocket.CLOSING && ws?.readyState !== WebSocket.CLOSED) {\n ws?.close();\n }\n ws = null;\n isClosing = false;\n }\n\n function isRecording(): boolean {\n return isRecordingFlag;\n }\n\n function isConnected(): boolean {\n return ws !== null && ws.readyState === WebSocket.OPEN;\n }\n\n return {\n connect,\n startRecording,\n stopRecording,\n close,\n isRecording,\n isConnected,\n };\n}\n\nexport default (authConfig: Pick<ASRClientConfig, \"getAccessToken\">) => (config: ASRClientConfig) =>\n createASRClient({ ...authConfig, ...config });\n\nexport type {\n SessionConfig,\n TurnDetectionConfig,\n InputAudioTranscriptionConfig,\n ClientEvent,\n ServerEvent,\n SessionUpdateEvent,\n InputAudioBufferAppendEvent,\n InputAudioBufferCommitEvent,\n SessionFinishEvent,\n SessionCreatedEvent,\n SessionUpdatedEvent,\n TranscriptionTextEvent,\n TranscriptionCompletedEvent,\n SessionFinishedEvent,\n ErrorEvent,\n};\n","/**\n * HTTP ASR Client - Press-to-talk style speech recognition\n *\n * HTTP-based speech recognition suitable for press-to-talk scenarios where you hold to speak\n * and release to recognize. Good for voice messages, voice search, etc.\n *\n * @example\n * ```typescript\n * const client = createASRHttpClient({\n * onRecordingStart() {\n * console.log(\"Recording started\");\n * },\n * onRecordingStop() {\n * console.log(\"Recording stopped\");\n * },\n * onResult(text) {\n * console.log(\"Recognized:\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * // Hold to speak, release to recognize\n * await client.startRecording();\n * // ... stop ...\n * const result = await client.stopRecording();\n * ```\n */\n\nimport { createHttpClient, type HttpClient } from \"@amaster.ai/http-client\";\n\nconst ASR_HTTP_PATH =\n \"/api/proxy/builtin/platform/qwen-asr/compatible-mode/v1/chat/completions\";\n\n// 录音内核\nconst RECORDER_WORKLET = `\nclass RecorderProcessor extends AudioWorkletProcessor {\n process(inputs) {\n const input = inputs[0];\n if (input && input[0]) {\n this.port.postMessage(input[0].slice(0));\n }\n return true;\n }\n}\nregisterProcessor('recorder-processor', RecorderProcessor);\n`;\n\ntype Status = \"idle\" | \"recording\" | \"recognizing\";\n\ninterface Recorder {\n /** Start recording */\n start(): Promise<void>;\n /**\n * Stop recording and get base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @returns Base64-encoded WAV audio data\n */\n stop(): Promise<void>;\n}\n\ninterface RecorderOptions {\n /** Called when recording starts */\n onStart?: () => void;\n /**\n * Called when recording stops, with base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @param base64 - Base64-encoded WAV audio data\n * @returns void\n */\n onStop?: (base64: string) => void;\n onError?: (error: Error) => void;\n}\n\n// 创建录音器\nasync function createWebRecorder(props?: RecorderOptions): Promise<Recorder> {\n let stream: MediaStream;\n let ctx: AudioContext;\n let node: AudioWorkletNode;\n let source: MediaStreamAudioSourceNode;\n const chunks: Int16Array[] = [];\n\n const cleanup = () => {\n try {\n source?.disconnect();\n node?.disconnect();\n stream?.getTracks().forEach((t) => t.stop());\n ctx?.close();\n } catch (e) {}\n };\n\n return {\n async start() {\n try {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: {\n channelCount: 1,\n echoCancellation: true,\n noiseSuppression: true,\n autoGainControl: true,\n },\n });\n\n ctx = new AudioContext();\n\n const blob = new Blob([RECORDER_WORKLET], {\n type: \"application/javascript\",\n });\n const url = URL.createObjectURL(blob);\n await ctx.audioWorklet.addModule(url);\n URL.revokeObjectURL(url);\n\n source = ctx.createMediaStreamSource(stream);\n node = new AudioWorkletNode(ctx, \"recorder-processor\");\n\n node.port.onmessage = (e) => {\n const input = e.data as Float32Array;\n const pcm = new Int16Array(input.length);\n for (let i = 0; i < input.length; i++) {\n const s = Math.max(-1, Math.min(1, input[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n chunks.push(pcm);\n };\n\n source.connect(node);\n\n props?.onStart?.();\n } catch (error) {\n props?.onError?.(\n error instanceof Error ? error : new Error(String(error)),\n );\n cleanup();\n }\n },\n\n async stop() {\n cleanup();\n\n const total = chunks.reduce((s, c) => s + c.length, 0);\n const pcm = new Int16Array(total);\n let offset = 0;\n for (const c of chunks) {\n pcm.set(c, offset);\n offset += c.length;\n }\n\n const result = { pcm, sampleRate: ctx?.sampleRate ?? 16000 };\n const base64 = await blobToBase64(\n pcmToWav(result.pcm, result.sampleRate),\n );\n props?.onStop?.(base64);\n },\n };\n}\n\n/**\n * Convert PCM to WAV Blob\n * @param pcm - PCM data\n * @param sampleRate - Sample rate\n * @returns WAV Blob\n */\nfunction pcmToWav(pcm: Int16Array, sampleRate: number): Blob {\n const buffer = new ArrayBuffer(44 + pcm.length * 2);\n const view = new DataView(buffer);\n\n const write = (o: number, s: string) => {\n for (let i = 0; i < s.length; i++) view.setUint8(o + i, s.charCodeAt(i));\n };\n\n write(0, \"RIFF\");\n view.setUint32(4, 36 + pcm.length * 2, true);\n write(8, \"WAVE\");\n write(12, \"fmt \");\n view.setUint32(16, 16, true);\n view.setUint16(20, 1, true);\n view.setUint16(22, 1, true);\n view.setUint32(24, sampleRate, true);\n view.setUint32(28, sampleRate * 2, true);\n view.setUint16(32, 2, true);\n view.setUint16(34, 16, true);\n write(36, \"data\");\n view.setUint32(40, pcm.length * 2, true);\n\n for (let i = 0; i < pcm.length; i++) {\n view.setInt16(44 + i * 2, pcm[i] || 0, true);\n }\n\n 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+ {"version":3,"sources":["../src/asr-client.ts","../src/http-asr-client.ts"],"sourcesContent":["/**\n * ASR Realtime WebSocket Client for Qwen-ASR Realtime API\n *\n * WebSocket-based real-time speech recognition for streaming transcription.\n * Follows the Qwen-ASR Realtime API protocol with proper event handling.\n *\n * @example\n * ```typescript\n * const client = createASRClient({\n * language: \"zh\",\n * enableVAD: true,\n * onReady() {\n * console.log(\"ASR connected\");\n * },\n * onTranscript(text, isFinal) {\n * console.log(isFinal ? \"[Final]\" : \"[Interim]\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * await client.connect();\n * await client.startRecording();\n * // ... stop ...\n * await client.stopRecording();\n * await client.close();\n * ```\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\n\nexport type ASRLanguage =\n | \"zh\" // 中文（普通话、四川话、闽南语、吴语）\n | \"yue\" // 粤语\n | \"en\" // 英文\n | \"ja\" // 日语\n | \"de\" // 德语\n | \"ko\" // 韩语\n | \"ru\" // 俄语\n | \"fr\" // 法语\n | \"pt\" // 葡萄牙语\n | \"ar\" // 阿拉伯语\n | \"it\" // 意大利语\n | \"es\" // 西班牙语\n | \"hi\" // 印地语\n | \"id\" // 印尼语\n | \"th\" // 泰语\n | \"tr\" // 土耳其语\n | \"uk\" // 乌克兰语\n | \"vi\" // 越南语\n | \"cs\" // 捷克语\n | \"da\" // 丹麦语\n | \"fil\" // 菲律宾语\n | \"fi\" // 芬兰语\n | \"is\" // 冰岛语\n | \"ms\" // 马来语\n | \"no\" // 挪威语\n | \"pl\" // 波兰语\n | \"sv\"; // 瑞典语\n\n// Client event types (sent to server)\ntype ClientEventType =\n | \"session.update\"\n | \"input_audio_buffer.append\"\n | \"input_audio_buffer.commit\"\n | \"session.finish\";\n\n// Server event types (received from server)\ntype ServerEventType =\n | \"session.created\"\n | \"session.updated\"\n | \"input_audio_buffer.speech_started\"\n | \"input_audio_buffer.speech_stopped\"\n | \"input_audio_buffer.committed\"\n | \"conversation.item.input_audio_transcription.text\"\n | \"conversation.item.input_audio_transcription.completed\"\n | \"session.finished\"\n | \"error\";\n\n// Base event interface\ninterface BaseEvent {\n event_id: string;\n type: ClientEventType | ServerEventType;\n}\n\n// Client events\ninterface SessionUpdateEvent extends BaseEvent {\n type: \"session.update\";\n session: SessionConfig;\n}\n\ninterface InputAudioBufferAppendEvent extends BaseEvent {\n type: \"input_audio_buffer.append\";\n audio: string;\n}\n\ninterface InputAudioBufferCommitEvent extends BaseEvent {\n type: \"input_audio_buffer.commit\";\n}\n\ninterface SessionFinishEvent extends BaseEvent {\n type: \"session.finish\";\n}\n\ntype ClientEvent =\n | SessionUpdateEvent\n | InputAudioBufferAppendEvent\n | InputAudioBufferCommitEvent\n | SessionFinishEvent;\n\n// Server events\ninterface SessionCreatedEvent extends BaseEvent {\n type: \"session.created\";\n session: { id: string };\n}\n\ninterface SessionUpdatedEvent extends BaseEvent {\n type: \"session.updated\";\n session: SessionConfig;\n}\n\ninterface SpeechStartedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_started\";\n}\n\ninterface SpeechStoppedEvent extends BaseEvent {\n type: \"input_audio_buffer.speech_stopped\";\n}\n\ninterface InputAudioBufferCommittedEvent extends BaseEvent {\n type: \"input_audio_buffer.committed\";\n}\n\ninterface TranscriptionTextEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.text\";\n text?: string;\n stash?: string;\n transcript?: string;\n}\n\ninterface TranscriptionCompletedEvent extends BaseEvent {\n type: \"conversation.item.input_audio_transcription.completed\";\n text?: string;\n transcript?: string;\n}\n\ninterface SessionFinishedEvent extends BaseEvent {\n type: \"session.finished\";\n}\n\ninterface ErrorEvent extends BaseEvent {\n type: \"error\";\n error: {\n message: string;\n code?: string;\n };\n}\n\ntype ServerEvent =\n | SessionCreatedEvent\n | SessionUpdatedEvent\n | SpeechStartedEvent\n | SpeechStoppedEvent\n | InputAudioBufferCommittedEvent\n | TranscriptionTextEvent\n | TranscriptionCompletedEvent\n | SessionFinishedEvent\n | ErrorEvent;\n\n// Session configuration\ninterface TurnDetectionConfig {\n type: \"server_vad\";\n /** VAD检测阈值，推荐设为 0.0，默认值 0.2，范围 [-1, 1] */\n threshold?: number;\n /** VAD断句检测阈值(ms)，推荐设为 400，默认值 800，范围 [200, 6000] */\n silence_duration_ms?: number;\n}\n\ninterface InputAudioTranscriptionConfig {\n language?: ASRLanguage;\n}\n\ninterface SessionConfig {\n input_audio_format?: \"pcm\" | \"opus\";\n sample_rate?: 16000 | 8000;\n input_audio_transcription?: InputAudioTranscriptionConfig;\n turn_detection?: TurnDetectionConfig | null;\n}\n\ninterface RealtimeRecorder {\n start(onAudio: (base64: string) => void): Promise<void>;\n stop(): Promise<void>;\n}\n\nasync function createRealtimeRecorder(): Promise<RealtimeRecorder> {\n let stream: MediaStream | null = null;\n let ctx: AudioContext | null = null;\n let source: MediaStreamAudioSourceNode | null = null;\n let processor: ScriptProcessorNode | null = null;\n\n if (!navigator.mediaDevices || !navigator.mediaDevices.getUserMedia) {\n throw new Error(\"未找到麦克风或无权限\");\n }\n\n return {\n async start(onAudio) {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: { sampleRate: 16000, channelCount: 1, echoCancellation: true },\n });\n\n log(\"✅ 麦克风已启动\", \"success\");\n log(\"💬 请对着麦克风说话，实时识别中...\", \"success\");\n\n ctx = new AudioContext({ sampleRate: 16000 });\n source = ctx.createMediaStreamSource(stream);\n processor = ctx.createScriptProcessor(4096, 1, 1);\n\n processor.onaudioprocess = (e) => {\n const inputData = e.inputBuffer.getChannelData(0);\n const pcm = new Int16Array(inputData.length);\n for (let i = 0; i < inputData.length; i++) {\n const s = Math.max(-1, Math.min(1, inputData[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n\n const bytes = new Uint8Array(pcm.buffer);\n let binary = \"\";\n for (let i = 0; i < bytes.length; i++) {\n binary += String.fromCharCode(bytes[i] || 0);\n }\n\n onAudio(btoa(binary));\n };\n\n source.connect(processor);\n processor.connect(ctx.destination);\n },\n\n async stop() {\n stream?.getTracks().forEach((t) => t.stop());\n source?.disconnect();\n processor?.disconnect();\n if (ctx) {\n await ctx.close();\n }\n stream = null;\n ctx = null;\n source = null;\n processor = null;\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\n// Generate unique event ID\nlet eventIdCounter = 0;\nfunction generateEventId(): string {\n return `event_${Date.now()}_${++eventIdCounter}`;\n}\n\nexport interface ASRClientConfig {\n /**\n * Audio format\n * @default \"pcm\"\n */\n audioFormat?: \"pcm\" | \"opus\";\n /**\n * Sample rate in Hz\n * @default 16000\n * @description 支持 16000 和 8000。设置为 8000 时，服务端会先升采样到16000Hz再进行识别，可能引入微小延迟。\n */\n sampleRate?: 16000 | 8000;\n /**\n * Audio source language\n * @default \"zh\"\n * @description 支持多种语言，包括 zh（中文）、yue（粤语）、en（英文）、ja（日语）等\n */\n language?: ASRLanguage;\n /**\n * Enable VAD (Voice Activity Detection) mode\n * @default true\n * @description true = VAD模式（服务端自动检测语音开始/结束），false = Manual模式（客户端手动控制）\n */\n enableVAD?: boolean;\n /**\n * VAD detection threshold\n * @default 0.2\n * @description 推荐设为 0.0。取值范围 [-1, 1]。较低的阈值会提高 VAD 的灵敏度。\n */\n vadThreshold?: number;\n /**\n * VAD silence duration threshold in milliseconds\n * @default 800\n * @description 推荐设为 400。取值范围 [200, 6000]。静音持续时长超过该阈值将被认为是语句结束。\n */\n vadSilenceDurationMs?: number;\n /**\n * Get access token for WebSocket authentication\n */\n getAccessToken?: () => string | null;\n /**\n * Called when connection is ready (session.created received and session.update sent)\n */\n onReady?: () => void;\n /**\n * Called when speech is detected (VAD mode only)\n */\n onSpeechStart?: () => void;\n /**\n * Called when speech stops (VAD mode only)\n */\n onSpeechEnd?: () => void;\n /**\n * Called on transcript result\n * @param text - Transcribed text\n * @param isFinal - Whether this is the final result\n */\n onTranscript?: (text: string, isFinal: boolean) => void;\n /**\n * Called when audio buffer is committed (non-VAD mode only)\n */\n onAudioBufferCommitted?: () => void;\n /**\n * Called when session is finished\n */\n onSessionFinished?: () => void;\n /**\n * Called on error\n */\n onError?: (error: Error) => void;\n /**\n * Called on close\n */\n onClose?: () => void;\n}\n\nexport interface ASRClient {\n /** Connect to ASR service and establish session */\n connect(): Promise<void>;\n /** Start recording from microphone */\n startRecording(): Promise<void>;\n /**\n * Stop recording\n * @description In non-VAD mode, this triggers recognition by sending input_audio_buffer.commit\n */\n stopRecording(): Promise<void>;\n /**\n * Close connection gracefully\n * @description Sends session.finish and waits for session.finished before closing\n */\n close(): Promise<void>;\n /**\n * Check if currently recording\n */\n isRecording(): boolean;\n /**\n * Check if connected to server\n */\n isConnected(): boolean;\n}\n\nfunction createASRClient(config: ASRClientConfig): ASRClient {\n const {\n onReady,\n onSpeechStart,\n onSpeechEnd,\n onTranscript,\n onAudioBufferCommitted,\n onSessionFinished,\n onError,\n onClose,\n getAccessToken,\n audioFormat = \"pcm\",\n sampleRate = 16000,\n language = \"zh\",\n enableVAD = true,\n vadThreshold = 0.2,\n vadSilenceDurationMs = 400,\n } = config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\n let isRecordingFlag = false;\n let isClosing = false;\n\n const path = ASR_PATH;\n\n /**\n * Send event to server\n */\n function sendEvent(event: ClientEvent): void {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n ws.send(JSON.stringify(event));\n }\n\n /**\n * Build session configuration\n */\n function buildSessionConfig(): SessionConfig {\n const sessionConfig: SessionConfig = {\n input_audio_format: audioFormat,\n sample_rate: sampleRate,\n input_audio_transcription: {\n language,\n },\n };\n\n if (enableVAD) {\n sessionConfig.turn_detection = {\n type: \"server_vad\",\n threshold: vadThreshold,\n silence_duration_ms: vadSilenceDurationMs,\n };\n } else {\n // Non-VAD mode: set turn_detection to null\n sessionConfig.turn_detection = null;\n }\n\n return sessionConfig;\n }\n\n /**\n * Send session.update event\n */\n function sendSessionUpdate(): void {\n const event: SessionUpdateEvent = {\n event_id: generateEventId(),\n type: \"session.update\",\n session: buildSessionConfig(),\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.append event\n */\n function sendAudioBufferAppend(audio: string): void {\n const event: InputAudioBufferAppendEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.append\",\n audio,\n };\n sendEvent(event);\n }\n\n /**\n * Send input_audio_buffer.commit event (for non-VAD mode)\n */\n function sendAudioBufferCommit(): void {\n const event: InputAudioBufferCommitEvent = {\n event_id: generateEventId(),\n type: \"input_audio_buffer.commit\",\n };\n sendEvent(event);\n }\n\n /**\n * Send session.finish event\n */\n function sendSessionFinish(): void {\n const event: SessionFinishEvent = {\n event_id: generateEventId(),\n type: \"session.finish\",\n };\n sendEvent(event);\n }\n\n /**\n * Handle server events\n */\n function handleServerEvent(data: ServerEvent): void {\n switch (data.type) {\n case \"session.created\":\n // Send session.update immediately after session.created\n try {\n sendSessionUpdate();\n } catch (err) {\n onError?.(\n new Error(\n \"Failed to send session.update: \" + (err instanceof Error ? err.message : String(err))\n )\n );\n }\n break;\n\n case \"session.updated\":\n // Server confirmed session configuration\n onReady?.();\n break;\n\n case \"input_audio_buffer.speech_started\":\n onSpeechStart?.();\n break;\n\n case \"input_audio_buffer.speech_stopped\":\n onSpeechEnd?.();\n break;\n\n case \"input_audio_buffer.committed\":\n onAudioBufferCommitted?.();\n break;\n\n case \"conversation.item.input_audio_transcription.text\":\n onTranscript?.(data.text || data.stash || data.transcript || \"\", false);\n break;\n\n case \"conversation.item.input_audio_transcription.completed\":\n onTranscript?.(data.text || data.transcript || \"\", true);\n break;\n\n case \"session.finished\":\n onSessionFinished?.();\n // Close WebSocket after receiving session.finished\n close();\n break;\n\n case \"error\":\n const err = new Error(data.error?.message || \"ASR error\");\n onError?.(err);\n break;\n\n default:\n // Unknown event type\n console.warn(\"[ASR] Unknown server event:\", (data as ServerEvent).type);\n }\n }\n\n async function connect(): Promise<void> {\n // Build WebSocket URL with optional token parameter\n let wsUrl = path;\n if (getAccessToken) {\n const token = getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n wsUrl = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n // Support both ws:// and wss:// protocols\n if (typeof window !== \"undefined\" && window.location) {\n const protocol = window.location.protocol === \"https:\" ? \"wss:\" : \"ws:\";\n if (!wsUrl.startsWith(\"ws://\") && !wsUrl.startsWith(\"wss://\")) {\n wsUrl = `${protocol}//${window.location.host}${wsUrl}`;\n }\n }\n\n ws = new WebSocket(wsUrl);\n\n return new Promise((resolve, reject) => {\n if (!ws) {\n reject(new Error(\"Failed to create WebSocket\"));\n return;\n }\n\n ws.onopen = () => {\n log(\"WebSocket connected\", \"success\");\n };\n\n ws.onmessage = (event) => {\n try {\n const data = JSON.parse(event.data) as ServerEvent;\n handleServerEvent(data);\n\n // Resolve on session.updated (ready state)\n if (data.type === \"session.updated\") {\n resolve();\n }\n } catch (err) {\n const error = new Error(\n \"Failed to parse server message: \" + (err instanceof Error ? err.message : String(err))\n );\n onError?.(error);\n reject(error);\n }\n };\n\n ws.onerror = (error) => {\n console.error(\"WebSocket error:\", error);\n const err = new Error(\"WebSocket error\");\n onError?.(err);\n reject(err);\n };\n\n ws.onclose = () => {\n isRecordingFlag = false;\n ws = null;\n onClose?.();\n };\n });\n }\n\n async function startRecording(): Promise<void> {\n if (!ws || ws.readyState !== WebSocket.OPEN) {\n throw new Error(\"WebSocket not connected\");\n }\n\n if (isRecordingFlag) {\n return;\n }\n\n try {\n recorder = await createRealtimeRecorder();\n isRecordingFlag = true;\n\n await recorder.start((audio) => {\n if (!ws || ws.readyState !== WebSocket.OPEN) return;\n try {\n sendAudioBufferAppend(audio);\n } catch (err) {\n console.error(\"[ASR] Failed to send audio:\", err);\n }\n });\n } catch (err) {\n console.error(\"[ASR] Failed to start recorder:\", err);\n onError?.(err instanceof Error ? err : new Error(String(err)));\n throw err;\n }\n }\n\n async function stopRecording(): Promise<void> {\n if (!isRecordingFlag) {\n return;\n }\n\n try {\n await recorder?.stop();\n } catch (err) {\n console.error(\"[ASR] Error stopping recorder:\", err);\n }\n recorder = null;\n isRecordingFlag = false;\n\n // In non-VAD mode, send commit to trigger recognition\n if (!enableVAD && ws?.readyState === WebSocket.OPEN) {\n try {\n sendAudioBufferCommit();\n } catch (err) {\n console.error(\"[ASR] Failed to send commit:\", err);\n }\n }\n }\n\n async function close(): Promise<void> {\n if (isClosing) {\n return;\n }\n\n isClosing = true;\n\n // Stop recording first\n await stopRecording();\n\n // Send session.finish if connected\n if (ws?.readyState === WebSocket.OPEN) {\n try {\n sendSessionFinish();\n // Wait a bit for session.finished response\n await new Promise((resolve) => setTimeout(resolve, 1000));\n } catch (err) {\n console.error(\"[ASR] Failed to send session.finish:\", err);\n }\n }\n\n // Close WebSocket\n if (ws && ws?.readyState !== WebSocket.CLOSING && ws?.readyState !== WebSocket.CLOSED) {\n ws?.close();\n }\n ws = null;\n isClosing = false;\n }\n\n function isRecording(): boolean {\n return isRecordingFlag;\n }\n\n function isConnected(): boolean {\n return ws !== null && ws.readyState === WebSocket.OPEN;\n }\n\n return {\n connect,\n startRecording,\n stopRecording,\n close,\n isRecording,\n isConnected,\n };\n}\n\nexport default (authConfig: Pick<ASRClientConfig, \"getAccessToken\">) => (config: ASRClientConfig) =>\n createASRClient({ ...authConfig, ...config });\n\nexport type {\n BaseEvent,\n ClientEventType,\n SessionConfig,\n TurnDetectionConfig,\n InputAudioTranscriptionConfig,\n ClientEvent,\n InputAudioBufferCommittedEvent,\n ServerEventType,\n ServerEvent,\n SessionUpdateEvent,\n InputAudioBufferAppendEvent,\n InputAudioBufferCommitEvent,\n SessionFinishEvent,\n SessionCreatedEvent,\n SessionUpdatedEvent,\n SpeechStartedEvent,\n SpeechStoppedEvent,\n TranscriptionTextEvent,\n TranscriptionCompletedEvent,\n SessionFinishedEvent,\n ErrorEvent,\n};\n","/**\n * HTTP ASR Client - Press-to-talk style speech recognition\n *\n * HTTP-based speech recognition suitable for press-to-talk scenarios where you hold to speak\n * and release to recognize. Good for voice messages, voice search, etc.\n *\n * @example\n * ```typescript\n * const client = createASRHttpClient({\n * onRecordingStart() {\n * console.log(\"Recording started\");\n * },\n * onRecordingStop() {\n * console.log(\"Recording stopped\");\n * },\n * onResult(text) {\n * console.log(\"Recognized:\", text);\n * },\n * onError(err) {\n * console.error(\"ASR error:\", err);\n * },\n * });\n *\n * // Hold to speak, release to recognize\n * await client.startRecording();\n * // ... stop ...\n * const result = await client.stopRecording();\n * ```\n */\n\nimport { createHttpClient, type HttpClient } from \"@amaster.ai/http-client\";\n\nconst ASR_HTTP_PATH =\n \"/api/proxy/builtin/platform/qwen-asr/compatible-mode/v1/chat/completions\";\n\n// 录音内核\nconst RECORDER_WORKLET = `\nclass RecorderProcessor extends AudioWorkletProcessor {\n process(inputs) {\n const input = inputs[0];\n if (input && input[0]) {\n this.port.postMessage(input[0].slice(0));\n }\n return true;\n }\n}\nregisterProcessor('recorder-processor', RecorderProcessor);\n`;\n\ntype Status = \"idle\" | \"recording\" | \"recognizing\";\n\nexport interface Recorder {\n /** Start recording */\n start(): Promise<void>;\n /**\n * Stop recording and get base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @returns Base64-encoded WAV audio data\n */\n stop(): Promise<void>;\n}\n\nexport interface RecorderOptions {\n /** Called when recording starts */\n onStart?: () => void;\n /**\n * Called when recording stops, with base64-encoded WAV audio data. You can use this data to call the ASR API.\n *\n * @param base64 - Base64-encoded WAV audio data\n * @returns void\n */\n onStop?: (base64: string) => void;\n onError?: (error: Error) => void;\n}\n\n// 创建录音器\nasync function createWebRecorder(props?: RecorderOptions): Promise<Recorder> {\n let stream: MediaStream;\n let ctx: AudioContext;\n let node: AudioWorkletNode;\n let source: MediaStreamAudioSourceNode;\n const chunks: Int16Array[] = [];\n\n const cleanup = () => {\n try {\n source?.disconnect();\n node?.disconnect();\n stream?.getTracks().forEach((t) => t.stop());\n ctx?.close();\n } catch (e) {}\n };\n\n return {\n async start() {\n try {\n stream = await navigator.mediaDevices.getUserMedia({\n audio: {\n channelCount: 1,\n echoCancellation: true,\n noiseSuppression: true,\n autoGainControl: true,\n },\n });\n\n ctx = new AudioContext();\n\n const blob = new Blob([RECORDER_WORKLET], {\n type: \"application/javascript\",\n });\n const url = URL.createObjectURL(blob);\n await ctx.audioWorklet.addModule(url);\n URL.revokeObjectURL(url);\n\n source = ctx.createMediaStreamSource(stream);\n node = new AudioWorkletNode(ctx, \"recorder-processor\");\n\n node.port.onmessage = (e) => {\n const input = e.data as Float32Array;\n const pcm = new Int16Array(input.length);\n for (let i = 0; i < input.length; i++) {\n const s = Math.max(-1, Math.min(1, input[i] || 0));\n pcm[i] = s < 0 ? s * 32768 : s * 32767;\n }\n chunks.push(pcm);\n };\n\n source.connect(node);\n\n props?.onStart?.();\n } catch (error) {\n props?.onError?.(\n error instanceof Error ? error : new Error(String(error)),\n );\n cleanup();\n }\n },\n\n async stop() {\n cleanup();\n\n const total = chunks.reduce((s, c) => s + c.length, 0);\n const pcm = new Int16Array(total);\n let offset = 0;\n for (const c of chunks) {\n pcm.set(c, offset);\n offset += c.length;\n }\n\n const result = { pcm, sampleRate: ctx?.sampleRate ?? 16000 };\n const base64 = await blobToBase64(\n pcmToWav(result.pcm, result.sampleRate),\n );\n props?.onStop?.(base64);\n },\n };\n}\n\n/**\n * Convert PCM to WAV Blob\n * @param pcm - PCM data\n * @param sampleRate - Sample rate\n * @returns WAV Blob\n */\nfunction pcmToWav(pcm: Int16Array, sampleRate: number): Blob {\n const buffer = new ArrayBuffer(44 + pcm.length * 2);\n const view = new DataView(buffer);\n\n const write = (o: number, s: string) => {\n for (let i = 0; i < s.length; i++) view.setUint8(o + i, s.charCodeAt(i));\n };\n\n write(0, \"RIFF\");\n view.setUint32(4, 36 + pcm.length * 2, true);\n write(8, \"WAVE\");\n write(12, \"fmt \");\n view.setUint32(16, 16, true);\n view.setUint16(20, 1, true);\n view.setUint16(22, 1, true);\n view.setUint32(24, sampleRate, true);\n view.setUint32(28, sampleRate * 2, true);\n view.setUint16(32, 2, true);\n view.setUint16(34, 16, true);\n write(36, \"data\");\n view.setUint32(40, pcm.length * 2, true);\n\n for (let i = 0; i < pcm.length; i++) {\n view.setInt16(44 + i * 2, pcm[i] || 0, true);\n }\n\n 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package/package.json CHANGED
@@ -1,6 +1,6 @@
1
1
  {
2
2
  "name": "@amaster.ai/asr-client",
3
- "version": "1.1.3",
3
+ "version": "1.1.5",
4
4
  "description": "Qwen ASR Realtime WebSocket client with microphone recording",
5
5
  "type": "module",
6
6
  "main": "./dist/index.cjs",
@@ -33,7 +33,7 @@
33
33
  "registry": "https://registry.npmjs.org/"
34
34
  },
35
35
  "dependencies": {
36
- "@amaster.ai/http-client": "1.1.3"
36
+ "@amaster.ai/http-client": "1.1.5"
37
37
  },
38
38
  "devDependencies": {
39
39
  "jsdom": "^23.2.0",