@amaster.ai/asr-client 1.0.0-beta.1

package/LICENSE

@@ -0,0 +1,21 @@
+MIT License
+
+Copyright (c) 2026 Amaster Team
+
+Permission is hereby granted, free of charge, to any person obtaining a copy
+of this software and associated documentation files (the "Software"), to deal
+in the Software without restriction, including without limitation the rights
+to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
+copies of the Software, and to permit persons to whom the Software is
+furnished to do so, subject to the following conditions:
+
+The above copyright notice and this permission notice shall be included in all
+copies or substantial portions of the Software.
+
+THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+SOFTWARE.

package/README.md

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+# @amaster.ai/asr-client
+
+通义千问实时语音识别（ASR）WebSocket 客户端，支持麦克风录音。
+
+## 特性
+
+- ✅ 完整的 WebSocket 协议封装
+- ✅ 自动麦克风采集和音频编码
+- ✅ 实时流式识别结果
+- ✅ VAD 自动检测语音开始/结束
+- ✅ TypeScript 类型支持
+- ✅ 基于实际调试验证的实现
+
+## 安装
+
+```bash
+npm install @amaster.ai/asr-client
+```
+
+## 快速开始
+
+```typescript
+import { createASRClient } from '@amaster.ai/asr-client';
+
+// 创建客户端
+const asr = createASRClient({
+  gatewayUrl: 'ws://www.appok.ai/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime',
+  language: 'zh',
+});
+
+// 监听事件
+asr.on('session-created', (session) => {
+  console.log('会话创建:', session.id, session.model);
+});
+
+asr.on('speech-started', () => {
+  console.log('🎤 检测到语音开始');
+});
+
+asr.on('transcript-partial', ({ text }) => {
+  console.log('⏳ 识别中:', text);
+});
+
+asr.on('transcript-final', ({ text }) => {
+  console.log('✅ 识别完成:', text);
+});
+
+// 连接并开始录音
+await asr.connect();
+await asr.startRecording(); // 自动请求麦克风权限
+
+// 停止录音
+asr.stopRecording();
+
+// 断开连接
+asr.disconnect();
+```
+
+## API 文档
+
+### createASRClient(config)
+
+创建 ASR 客户端实例。
+
+**参数**：
+- `gatewayUrl`: Gateway WebSocket URL（会自动追加 model 参数）
+- `language`: 识别语言，默认 `'zh'`
+- `audioFormat`: 音频格式，默认 `'pcm16'`
+- `sampleRate`: 采样率，默认 `16000`
+
+### ASRClient
+
+#### connect()
+
+连接到 ASR 服务。
+
+#### startRecording()
+
+开始录音识别（会请求麦克风权限）。
+
+#### stopRecording()
+
+停止录音。
+
+#### on(event, callback)
+
+监听事件。
+
+**事件类型**：
+- `connected`: WebSocket 连接建立
+- `session-created`: 会话创建成功
+- `recording-started`: 录音已开始
+- `speech-started`: VAD 检测到语音开始
+- `speech-stopped`: VAD 检测到语音停止
+- `transcript-partial`: 中间识别结果（实时更新）
+- `transcript-final`: 最终识别结果（确认）
+- `error`: 发生错误
+- `closed`: 连接关闭
+
+#### disconnect()
+
+停止录音并断开连接。
+
+## 实现细节
+
+### 音频采集
+
+- 采样率：16kHz
+- 格式：PCM 16-bit Mono
+- 编码：Base64
+- 发送频率：约每秒 4 次（每次 4096 采样点）
+
+### WebSocket 协议
+
+基于实际验证的消息类型：
+- `input_audio_buffer.append` - 发送音频数据
+
+### 识别结果
+
+实际事件类型（与文档不同！）：
+- `conversation.item.input_audio_transcription.text` - 中间结果
+- `conversation.item.input_audio_transcription.completed` - 最终结果
+
+### VAD（语音活动检测）
+
+服务器自动检测：
+- `input_audio_buffer.speech_started` - 开始说话
+- `input_audio_buffer.speech_stopped` - 停止说话
+
+## License
+
+MIT

package/dist/index.cjs

@@ -0,0 +1,162 @@
+var __defProp = Object.defineProperty;
+var __getOwnPropDesc = Object.getOwnPropertyDescriptor;
+var __getOwnPropNames = Object.getOwnPropertyNames;
+var __hasOwnProp = Object.prototype.hasOwnProperty;
+var __export = (target, all) => {
+  for (var name in all)
+    __defProp(target, name, { get: all[name], enumerable: true });
+};
+var __copyProps = (to, from, except, desc) => {
+  if (from && typeof from === "object" || typeof from === "function") {
+    for (let key of __getOwnPropNames(from))
+      if (!__hasOwnProp.call(to, key) && key !== except)
+        __defProp(to, key, { get: () => from[key], enumerable: !(desc = __getOwnPropDesc(from, key)) || desc.enumerable });
+  }
+  return to;
+};
+var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: true }), mod);
+
+// src/index.ts
+var index_exports = {};
+__export(index_exports, {
+  createASRClient: () => createASRClient
+});
+module.exports = __toCommonJS(index_exports);
+
+// src/asr-client.ts
+function createASRClient(config) {
+  const {
+    url,
+    audioFormat = "pcm16",
+    sampleRate = 16e3,
+    onReady,
+    onSpeechStart,
+    onSpeechEnd,
+    onTranscript,
+    onError
+  } = config;
+  let ws = null;
+  let mediaStream = null;
+  let audioContext = null;
+  let processor = null;
+  async function connect() {
+    return new Promise((resolve, reject) => {
+      ws = new WebSocket(url);
+      ws.onopen = () => {
+      };
+      ws.onmessage = (event) => {
+        var _a;
+        const data = JSON.parse(event.data);
+        if (data.type === "session.created") {
+          onReady == null ? void 0 : onReady();
+          resolve();
+        }
+        if (data.type === "input_audio_buffer.speech_started") {
+          onSpeechStart == null ? void 0 : onSpeechStart();
+        }
+        if (data.type === "input_audio_buffer.speech_stopped") {
+          onSpeechEnd == null ? void 0 : onSpeechEnd();
+        }
+        if (data.type === "conversation.item.input_audio_transcription.text") {
+          onTranscript == null ? void 0 : onTranscript(data.text || "", false);
+        }
+        if (data.type === "conversation.item.input_audio_transcription.completed") {
+          onTranscript == null ? void 0 : onTranscript(data.text || data.transcript || "", true);
+        }
+        if (data.type === "error") {
+          const err = new Error(((_a = data.error) == null ? void 0 : _a.message) || "Unknown error");
+          onError == null ? void 0 : onError(err);
+          reject(err);
+        }
+      };
+      ws.onerror = () => {
+        const err = new Error("WebSocket connection error");
+        onError == null ? void 0 : onError(err);
+        reject(err);
+      };
+      ws.onclose = () => {
+        ws = null;
+      };
+    });
+  }
+  async function startRecording() {
+    if (typeof window === "undefined") {
+      throw new Error("Recording only supported in browser");
+    }
+    if (!ws || ws.readyState !== WebSocket.OPEN) {
+      throw new Error("WebSocket not connected");
+    }
+    try {
+      mediaStream = await navigator.mediaDevices.getUserMedia({
+        audio: {
+          sampleRate,
+          channelCount: 1,
+          echoCancellation: true,
+          noiseSuppression: true
+        }
+      });
+      audioContext = new AudioContext({ sampleRate });
+      const source = audioContext.createMediaStreamSource(mediaStream);
+      processor = audioContext.createScriptProcessor(4096, 1, 1);
+      processor.onaudioprocess = (e) => {
+        if (!ws || ws.readyState !== WebSocket.OPEN) return;
+        const inputData = e.inputBuffer.getChannelData(0);
+        const pcm = new Int16Array(inputData.length);
+        for (let i = 0; i < inputData.length; i++) {
+          const s = Math.max(-1, Math.min(1, inputData[i]));
+          pcm[i] = s < 0 ? s * 32768 : s * 32767;
+        }
+        const bytes = new Uint8Array(pcm.buffer);
+        let binary = "";
+        for (let i = 0; i < bytes.length; i++) {
+          binary += String.fromCharCode(bytes[i]);
+        }
+        const base64 = btoa(binary);
+        ws.send(JSON.stringify({
+          type: "input_audio_buffer.append",
+          audio: base64
+        }));
+      };
+      source.connect(processor);
+      processor.connect(audioContext.destination);
+    } catch (err) {
+      onError == null ? void 0 : onError(err);
+      throw err;
+    }
+  }
+  function stopRecording() {
+    if (mediaStream) {
+      mediaStream.getTracks().forEach((track) => track.stop());
+      mediaStream = null;
+    }
+    if (processor) {
+      processor.disconnect();
+      processor = null;
+    }
+    if (audioContext) {
+      audioContext.close();
+      audioContext = null;
+    }
+    if (ws && ws.readyState === WebSocket.OPEN) {
+      ws.send(JSON.stringify({ type: "input_audio_buffer.commit" }));
+    }
+  }
+  function close() {
+    stopRecording();
+    if (ws) {
+      ws.close();
+      ws = null;
+    }
+  }
+  return {
+    connect,
+    startRecording,
+    stopRecording,
+    close
+  };
+}
+// Annotate the CommonJS export names for ESM import in node:
+0 && (module.exports = {
+  createASRClient
+});
+//# sourceMappingURL=index.cjs.map

package/dist/index.cjs.map

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+{"version":3,"sources":["../src/index.ts","../src/asr-client.ts"],"sourcesContent":["export type { ASRClient, ASRClientConfig } from './asr-client';\nexport { createASRClient } from './asr-client';\n","/**\n * ASR Realtime WebSocket Client\n */\n\nexport interface ASRClientConfig {\n  /** WebSocket endpoint URL */\n  url: string;\n  /** Audio format, default 'pcm16' */\n  audioFormat?: 'pcm16' | 'g711a' | 'g711u';\n  /** Sample rate, default 16000 */\n  sampleRate?: number;\n  /** Called when connection is ready */\n  onReady?: () => void;\n  /** Called when speech is detected */\n  onSpeechStart?: () => void;\n  /** Called when speech stops */\n  onSpeechEnd?: () => void;\n  /** Called on transcript result */\n  onTranscript?: (text: string, isFinal: boolean) => void;\n  /** Called on error */\n  onError?: (error: Error) => void;\n}\n\nexport interface ASRClient {\n  /** Connect to ASR service */\n  connect(): Promise<void>;\n  /** Start recording from microphone */\n  startRecording(): Promise<void>;\n  /** Stop recording */\n  stopRecording(): void;\n  /** Close connection */\n  close(): void;\n}\n\nexport function createASRClient(config: ASRClientConfig): ASRClient {\n  const {\n    url,\n    audioFormat = 'pcm16',\n    sampleRate = 16000,\n    onReady,\n    onSpeechStart,\n    onSpeechEnd,\n    onTranscript,\n    onError,\n  } = config;\n\n  let ws: WebSocket | null = null;\n  let mediaStream: MediaStream | null = null;\n  let audioContext: AudioContext | null = null;\n  let processor: ScriptProcessorNode | null = null;\n\n  async function connect(): Promise<void> {\n    return new Promise((resolve, reject) => {\n      ws = new WebSocket(url);\n\n      ws.onopen = () => {};\n\n      ws.onmessage = (event) => {\n        const data = JSON.parse(event.data);\n\n        if (data.type === 'session.created') {\n          onReady?.();\n          resolve();\n        }\n\n        if (data.type === 'input_audio_buffer.speech_started') {\n          onSpeechStart?.();\n        }\n\n        if (data.type === 'input_audio_buffer.speech_stopped') {\n          onSpeechEnd?.();\n        }\n\n        if (data.type === 'conversation.item.input_audio_transcription.text') {\n          onTranscript?.(data.text || '', false);\n        }\n\n        if (data.type === 'conversation.item.input_audio_transcription.completed') {\n          onTranscript?.(data.text || data.transcript || '', true);\n        }\n\n        if (data.type === 'error') {\n          const err = new Error(data.error?.message || 'Unknown error');\n          onError?.(err);\n          reject(err);\n        }\n      };\n\n      ws.onerror = () => {\n        const err = new Error('WebSocket connection error');\n        onError?.(err);\n        reject(err);\n      };\n\n      ws.onclose = () => {\n        ws = null;\n      };\n    });\n  }\n\n  async function startRecording(): Promise<void> {\n    if (typeof window === 'undefined') {\n      throw new Error('Recording only supported in browser');\n    }\n\n    if (!ws || ws.readyState !== WebSocket.OPEN) {\n      throw new Error('WebSocket not connected');\n    }\n\n    try {\n      mediaStream = await navigator.mediaDevices.getUserMedia({\n        audio: {\n          sampleRate,\n          channelCount: 1,\n          echoCancellation: true,\n          noiseSuppression: true,\n        },\n      });\n\n      audioContext = new AudioContext({ sampleRate });\n      const source = audioContext.createMediaStreamSource(mediaStream);\n      processor = audioContext.createScriptProcessor(4096, 1, 1);\n\n      processor.onaudioprocess = (e) => {\n        if (!ws || ws.readyState !== WebSocket.OPEN) return;\n\n        const inputData = e.inputBuffer.getChannelData(0);\n\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]));\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]);\n        }\n        const base64 = btoa(binary);\n\n        ws.send(JSON.stringify({\n          type: 'input_audio_buffer.append',\n          audio: base64,\n        }));\n      };\n\n      source.connect(processor);\n      processor.connect(audioContext.destination);\n    } catch (err) {\n      onError?.(err as Error);\n      throw err;\n    }\n  }\n\n  function stopRecording() {\n    if (mediaStream) {\n      mediaStream.getTracks().forEach(track => track.stop());\n      mediaStream = null;\n    }\n\n    if (processor) {\n      processor.disconnect();\n      processor = null;\n    }\n\n    if (audioContext) {\n      audioContext.close();\n      audioContext = null;\n    }\n\n    if (ws && ws.readyState === WebSocket.OPEN) {\n      ws.send(JSON.stringify({ type: 'input_audio_buffer.commit' }));\n    }\n  }\n\n  function close() {\n    stopRecording();\n    if (ws) {\n      ws.close();\n      ws = null;\n    }\n  }\n\n  return {\n    connect,\n    startRecording,\n    stopRecording,\n    close,\n  };\n}\n"],"mappings":";;;;;;;;;;;;;;;;;;;AAAA;AAAA;AAAA;AAAA;AAAA;;;ACkCO,SAAS,gBAAgB,QAAoC;AAClE,QAAM;AAAA,IACJ;AAAA,IACA,cAAc;AAAA,IACd,aAAa;AAAA,IACb;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,EACF,IAAI;AAEJ,MAAI,KAAuB;AAC3B,MAAI,cAAkC;AACtC,MAAI,eAAoC;AACxC,MAAI,YAAwC;AAE5C,iBAAe,UAAyB;AACtC,WAAO,IAAI,QAAQ,CAAC,SAAS,WAAW;AACtC,WAAK,IAAI,UAAU,GAAG;AAEtB,SAAG,SAAS,MAAM;AAAA,MAAC;AAEnB,SAAG,YAAY,CAAC,UAAU;AAzDhC;AA0DQ,cAAM,OAAO,KAAK,MAAM,MAAM,IAAI;AAElC,YAAI,KAAK,SAAS,mBAAmB;AACnC;AACA,kBAAQ;AAAA,QACV;AAEA,YAAI,KAAK,SAAS,qCAAqC;AACrD;AAAA,QACF;AAEA,YAAI,KAAK,SAAS,qCAAqC;AACrD;AAAA,QACF;AAEA,YAAI,KAAK,SAAS,oDAAoD;AACpE,uDAAe,KAAK,QAAQ,IAAI;AAAA,QAClC;AAEA,YAAI,KAAK,SAAS,yDAAyD;AACzE,uDAAe,KAAK,QAAQ,KAAK,cAAc,IAAI;AAAA,QACrD;AAEA,YAAI,KAAK,SAAS,SAAS;AACzB,gBAAM,MAAM,IAAI,QAAM,UAAK,UAAL,mBAAY,YAAW,eAAe;AAC5D,6CAAU;AACV,iBAAO,GAAG;AAAA,QACZ;AAAA,MACF;AAEA,SAAG,UAAU,MAAM;AACjB,cAAM,MAAM,IAAI,MAAM,4BAA4B;AAClD,2CAAU;AACV,eAAO,GAAG;AAAA,MACZ;AAEA,SAAG,UAAU,MAAM;AACjB,aAAK;AAAA,MACP;AAAA,IACF,CAAC;AAAA,EACH;AAEA,iBAAe,iBAAgC;AAC7C,QAAI,OAAO,WAAW,aAAa;AACjC,YAAM,IAAI,MAAM,qCAAqC;AAAA,IACvD;AAEA,QAAI,CAAC,MAAM,GAAG,eAAe,UAAU,MAAM;AAC3C,YAAM,IAAI,MAAM,yBAAyB;AAAA,IAC3C;AAEA,QAAI;AACF,oBAAc,MAAM,UAAU,aAAa,aAAa;AAAA,QACtD,OAAO;AAAA,UACL;AAAA,UACA,cAAc;AAAA,UACd,kBAAkB;AAAA,UAClB,kBAAkB;AAAA,QACpB;AAAA,MACF,CAAC;AAED,qBAAe,IAAI,aAAa,EAAE,WAAW,CAAC;AAC9C,YAAM,SAAS,aAAa,wBAAwB,WAAW;AAC/D,kBAAY,aAAa,sBAAsB,MAAM,GAAG,CAAC;AAEzD,gBAAU,iBAAiB,CAAC,MAAM;AAChC,YAAI,CAAC,MAAM,GAAG,eAAe,UAAU,KAAM;AAE7C,cAAM,YAAY,EAAE,YAAY,eAAe,CAAC;AAEhD,cAAM,MAAM,IAAI,WAAW,UAAU,MAAM;AAC3C,iBAAS,IAAI,GAAG,IAAI,UAAU,QAAQ,KAAK;AACzC,gBAAM,IAAI,KAAK,IAAI,IAAI,KAAK,IAAI,GAAG,UAAU,CAAC,CAAC,CAAC;AAChD,cAAI,CAAC,IAAI,IAAI,IAAI,IAAI,QAAQ,IAAI;AAAA,QACnC;AAEA,cAAM,QAAQ,IAAI,WAAW,IAAI,MAAM;AACvC,YAAI,SAAS;AACb,iBAAS,IAAI,GAAG,IAAI,MAAM,QAAQ,KAAK;AACrC,oBAAU,OAAO,aAAa,MAAM,CAAC,CAAC;AAAA,QACxC;AACA,cAAM,SAAS,KAAK,MAAM;AAE1B,WAAG,KAAK,KAAK,UAAU;AAAA,UACrB,MAAM;AAAA,UACN,OAAO;AAAA,QACT,CAAC,CAAC;AAAA,MACJ;AAEA,aAAO,QAAQ,SAAS;AACxB,gBAAU,QAAQ,aAAa,WAAW;AAAA,IAC5C,SAAS,KAAK;AACZ,yCAAU;AACV,YAAM;AAAA,IACR;AAAA,EACF;AAEA,WAAS,gBAAgB;AACvB,QAAI,aAAa;AACf,kBAAY,UAAU,EAAE,QAAQ,WAAS,MAAM,KAAK,CAAC;AACrD,oBAAc;AAAA,IAChB;AAEA,QAAI,WAAW;AACb,gBAAU,WAAW;AACrB,kBAAY;AAAA,IACd;AAEA,QAAI,cAAc;AAChB,mBAAa,MAAM;AACnB,qBAAe;AAAA,IACjB;AAEA,QAAI,MAAM,GAAG,eAAe,UAAU,MAAM;AAC1C,SAAG,KAAK,KAAK,UAAU,EAAE,MAAM,4BAA4B,CAAC,CAAC;AAAA,IAC/D;AAAA,EACF;AAEA,WAAS,QAAQ;AACf,kBAAc;AACd,QAAI,IAAI;AACN,SAAG,MAAM;AACT,WAAK;AAAA,IACP;AAAA,EACF;AAEA,SAAO;AAAA,IACL;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,EACF;AACF;","names":[]}

package/dist/index.d.cts

@@ -0,0 +1,34 @@
+/**
+ * ASR Realtime WebSocket Client
+ */
+interface ASRClientConfig {
+    /** WebSocket endpoint URL */
+    url: string;
+    /** Audio format, default 'pcm16' */
+    audioFormat?: 'pcm16' | 'g711a' | 'g711u';
+    /** Sample rate, default 16000 */
+    sampleRate?: number;
+    /** Called when connection is ready */
+    onReady?: () => void;
+    /** Called when speech is detected */
+    onSpeechStart?: () => void;
+    /** Called when speech stops */
+    onSpeechEnd?: () => void;
+    /** Called on transcript result */
+    onTranscript?: (text: string, isFinal: boolean) => void;
+    /** Called on error */
+    onError?: (error: Error) => void;
+}
+interface ASRClient {
+    /** Connect to ASR service */
+    connect(): Promise<void>;
+    /** Start recording from microphone */
+    startRecording(): Promise<void>;
+    /** Stop recording */
+    stopRecording(): void;
+    /** Close connection */
+    close(): void;
+}
+declare function createASRClient(config: ASRClientConfig): ASRClient;
+
+export { type ASRClient, type ASRClientConfig, createASRClient };

package/dist/index.d.ts

@@ -0,0 +1,34 @@
+/**
+ * ASR Realtime WebSocket Client
+ */
+interface ASRClientConfig {
+    /** WebSocket endpoint URL */
+    url: string;
+    /** Audio format, default 'pcm16' */
+    audioFormat?: 'pcm16' | 'g711a' | 'g711u';
+    /** Sample rate, default 16000 */
+    sampleRate?: number;
+    /** Called when connection is ready */
+    onReady?: () => void;
+    /** Called when speech is detected */
+    onSpeechStart?: () => void;
+    /** Called when speech stops */
+    onSpeechEnd?: () => void;
+    /** Called on transcript result */
+    onTranscript?: (text: string, isFinal: boolean) => void;
+    /** Called on error */
+    onError?: (error: Error) => void;
+}
+interface ASRClient {
+    /** Connect to ASR service */
+    connect(): Promise<void>;
+    /** Start recording from microphone */
+    startRecording(): Promise<void>;
+    /** Stop recording */
+    stopRecording(): void;
+    /** Close connection */
+    close(): void;
+}
+declare function createASRClient(config: ASRClientConfig): ASRClient;
+
+export { type ASRClient, type ASRClientConfig, createASRClient };

package/dist/index.js

@@ -0,0 +1,136 @@
+// src/asr-client.ts
+function createASRClient(config) {
+  const {
+    url,
+    audioFormat = "pcm16",
+    sampleRate = 16e3,
+    onReady,
+    onSpeechStart,
+    onSpeechEnd,
+    onTranscript,
+    onError
+  } = config;
+  let ws = null;
+  let mediaStream = null;
+  let audioContext = null;
+  let processor = null;
+  async function connect() {
+    return new Promise((resolve, reject) => {
+      ws = new WebSocket(url);
+      ws.onopen = () => {
+      };
+      ws.onmessage = (event) => {
+        var _a;
+        const data = JSON.parse(event.data);
+        if (data.type === "session.created") {
+          onReady == null ? void 0 : onReady();
+          resolve();
+        }
+        if (data.type === "input_audio_buffer.speech_started") {
+          onSpeechStart == null ? void 0 : onSpeechStart();
+        }
+        if (data.type === "input_audio_buffer.speech_stopped") {
+          onSpeechEnd == null ? void 0 : onSpeechEnd();
+        }
+        if (data.type === "conversation.item.input_audio_transcription.text") {
+          onTranscript == null ? void 0 : onTranscript(data.text || "", false);
+        }
+        if (data.type === "conversation.item.input_audio_transcription.completed") {
+          onTranscript == null ? void 0 : onTranscript(data.text || data.transcript || "", true);
+        }
+        if (data.type === "error") {
+          const err = new Error(((_a = data.error) == null ? void 0 : _a.message) || "Unknown error");
+          onError == null ? void 0 : onError(err);
+          reject(err);
+        }
+      };
+      ws.onerror = () => {
+        const err = new Error("WebSocket connection error");
+        onError == null ? void 0 : onError(err);
+        reject(err);
+      };
+      ws.onclose = () => {
+        ws = null;
+      };
+    });
+  }
+  async function startRecording() {
+    if (typeof window === "undefined") {
+      throw new Error("Recording only supported in browser");
+    }
+    if (!ws || ws.readyState !== WebSocket.OPEN) {
+      throw new Error("WebSocket not connected");
+    }
+    try {
+      mediaStream = await navigator.mediaDevices.getUserMedia({
+        audio: {
+          sampleRate,
+          channelCount: 1,
+          echoCancellation: true,
+          noiseSuppression: true
+        }
+      });
+      audioContext = new AudioContext({ sampleRate });
+      const source = audioContext.createMediaStreamSource(mediaStream);
+      processor = audioContext.createScriptProcessor(4096, 1, 1);
+      processor.onaudioprocess = (e) => {
+        if (!ws || ws.readyState !== WebSocket.OPEN) return;
+        const inputData = e.inputBuffer.getChannelData(0);
+        const pcm = new Int16Array(inputData.length);
+        for (let i = 0; i < inputData.length; i++) {
+          const s = Math.max(-1, Math.min(1, inputData[i]));
+          pcm[i] = s < 0 ? s * 32768 : s * 32767;
+        }
+        const bytes = new Uint8Array(pcm.buffer);
+        let binary = "";
+        for (let i = 0; i < bytes.length; i++) {
+          binary += String.fromCharCode(bytes[i]);
+        }
+        const base64 = btoa(binary);
+        ws.send(JSON.stringify({
+          type: "input_audio_buffer.append",
+          audio: base64
+        }));
+      };
+      source.connect(processor);
+      processor.connect(audioContext.destination);
+    } catch (err) {
+      onError == null ? void 0 : onError(err);
+      throw err;
+    }
+  }
+  function stopRecording() {
+    if (mediaStream) {
+      mediaStream.getTracks().forEach((track) => track.stop());
+      mediaStream = null;
+    }
+    if (processor) {
+      processor.disconnect();
+      processor = null;
+    }
+    if (audioContext) {
+      audioContext.close();
+      audioContext = null;
+    }
+    if (ws && ws.readyState === WebSocket.OPEN) {
+      ws.send(JSON.stringify({ type: "input_audio_buffer.commit" }));
+    }
+  }
+  function close() {
+    stopRecording();
+    if (ws) {
+      ws.close();
+      ws = null;
+    }
+  }
+  return {
+    connect,
+    startRecording,
+    stopRecording,
+    close
+  };
+}
+export {
+  createASRClient
+};
+//# sourceMappingURL=index.js.map

package/dist/index.js.map

@@ -0,0 +1 @@
+{"version":3,"sources":["../src/asr-client.ts"],"sourcesContent":["/**\n * ASR Realtime WebSocket Client\n */\n\nexport interface ASRClientConfig {\n  /** WebSocket endpoint URL */\n  url: string;\n  /** Audio format, default 'pcm16' */\n  audioFormat?: 'pcm16' | 'g711a' | 'g711u';\n  /** Sample rate, default 16000 */\n  sampleRate?: number;\n  /** Called when connection is ready */\n  onReady?: () => void;\n  /** Called when speech is detected */\n  onSpeechStart?: () => void;\n  /** Called when speech stops */\n  onSpeechEnd?: () => void;\n  /** Called on transcript result */\n  onTranscript?: (text: string, isFinal: boolean) => void;\n  /** Called on error */\n  onError?: (error: Error) => void;\n}\n\nexport interface ASRClient {\n  /** Connect to ASR service */\n  connect(): Promise<void>;\n  /** Start recording from microphone */\n  startRecording(): Promise<void>;\n  /** Stop recording */\n  stopRecording(): void;\n  /** Close connection */\n  close(): void;\n}\n\nexport function createASRClient(config: ASRClientConfig): ASRClient {\n  const {\n    url,\n    audioFormat = 'pcm16',\n    sampleRate = 16000,\n    onReady,\n    onSpeechStart,\n    onSpeechEnd,\n    onTranscript,\n    onError,\n  } = config;\n\n  let ws: WebSocket | null = null;\n  let mediaStream: MediaStream | null = null;\n  let audioContext: AudioContext | null = null;\n  let processor: ScriptProcessorNode | null = null;\n\n  async function connect(): Promise<void> {\n    return new Promise((resolve, reject) => {\n      ws = new WebSocket(url);\n\n      ws.onopen = () => {};\n\n      ws.onmessage = (event) => {\n        const data = JSON.parse(event.data);\n\n        if (data.type === 'session.created') {\n          onReady?.();\n          resolve();\n        }\n\n        if (data.type === 'input_audio_buffer.speech_started') {\n          onSpeechStart?.();\n        }\n\n        if (data.type === 'input_audio_buffer.speech_stopped') {\n          onSpeechEnd?.();\n        }\n\n        if (data.type === 'conversation.item.input_audio_transcription.text') {\n          onTranscript?.(data.text || '', false);\n        }\n\n        if (data.type === 'conversation.item.input_audio_transcription.completed') {\n          onTranscript?.(data.text || data.transcript || '', true);\n        }\n\n        if (data.type === 'error') {\n          const err = new Error(data.error?.message || 'Unknown error');\n          onError?.(err);\n          reject(err);\n        }\n      };\n\n      ws.onerror = () => {\n        const err = new Error('WebSocket connection error');\n        onError?.(err);\n        reject(err);\n      };\n\n      ws.onclose = () => {\n        ws = null;\n      };\n    });\n  }\n\n  async function startRecording(): Promise<void> {\n    if (typeof window === 'undefined') {\n      throw new Error('Recording only supported in browser');\n    }\n\n    if (!ws || ws.readyState !== WebSocket.OPEN) {\n      throw new Error('WebSocket not connected');\n    }\n\n    try {\n      mediaStream = await navigator.mediaDevices.getUserMedia({\n        audio: {\n          sampleRate,\n          channelCount: 1,\n          echoCancellation: true,\n          noiseSuppression: true,\n        },\n      });\n\n      audioContext = new AudioContext({ sampleRate });\n      const source = audioContext.createMediaStreamSource(mediaStream);\n      processor = audioContext.createScriptProcessor(4096, 1, 1);\n\n      processor.onaudioprocess = (e) => {\n        if (!ws || ws.readyState !== WebSocket.OPEN) return;\n\n        const inputData = e.inputBuffer.getChannelData(0);\n\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]));\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]);\n        }\n        const base64 = btoa(binary);\n\n        ws.send(JSON.stringify({\n          type: 'input_audio_buffer.append',\n          audio: base64,\n        }));\n      };\n\n      source.connect(processor);\n      processor.connect(audioContext.destination);\n    } catch (err) {\n      onError?.(err as Error);\n      throw err;\n    }\n  }\n\n  function stopRecording() {\n    if (mediaStream) {\n      mediaStream.getTracks().forEach(track => track.stop());\n      mediaStream = null;\n    }\n\n    if (processor) {\n      processor.disconnect();\n      processor = null;\n    }\n\n    if (audioContext) {\n      audioContext.close();\n      audioContext = null;\n    }\n\n    if (ws && ws.readyState === WebSocket.OPEN) {\n      ws.send(JSON.stringify({ type: 'input_audio_buffer.commit' }));\n    }\n  }\n\n  function close() {\n    stopRecording();\n    if (ws) {\n      ws.close();\n      ws = null;\n    }\n  }\n\n  return {\n    connect,\n    startRecording,\n    stopRecording,\n    close,\n  };\n}\n"],"mappings":";AAkCO,SAAS,gBAAgB,QAAoC;AAClE,QAAM;AAAA,IACJ;AAAA,IACA,cAAc;AAAA,IACd,aAAa;AAAA,IACb;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,EACF,IAAI;AAEJ,MAAI,KAAuB;AAC3B,MAAI,cAAkC;AACtC,MAAI,eAAoC;AACxC,MAAI,YAAwC;AAE5C,iBAAe,UAAyB;AACtC,WAAO,IAAI,QAAQ,CAAC,SAAS,WAAW;AACtC,WAAK,IAAI,UAAU,GAAG;AAEtB,SAAG,SAAS,MAAM;AAAA,MAAC;AAEnB,SAAG,YAAY,CAAC,UAAU;AAzDhC;AA0DQ,cAAM,OAAO,KAAK,MAAM,MAAM,IAAI;AAElC,YAAI,KAAK,SAAS,mBAAmB;AACnC;AACA,kBAAQ;AAAA,QACV;AAEA,YAAI,KAAK,SAAS,qCAAqC;AACrD;AAAA,QACF;AAEA,YAAI,KAAK,SAAS,qCAAqC;AACrD;AAAA,QACF;AAEA,YAAI,KAAK,SAAS,oDAAoD;AACpE,uDAAe,KAAK,QAAQ,IAAI;AAAA,QAClC;AAEA,YAAI,KAAK,SAAS,yDAAyD;AACzE,uDAAe,KAAK,QAAQ,KAAK,cAAc,IAAI;AAAA,QACrD;AAEA,YAAI,KAAK,SAAS,SAAS;AACzB,gBAAM,MAAM,IAAI,QAAM,UAAK,UAAL,mBAAY,YAAW,eAAe;AAC5D,6CAAU;AACV,iBAAO,GAAG;AAAA,QACZ;AAAA,MACF;AAEA,SAAG,UAAU,MAAM;AACjB,cAAM,MAAM,IAAI,MAAM,4BAA4B;AAClD,2CAAU;AACV,eAAO,GAAG;AAAA,MACZ;AAEA,SAAG,UAAU,MAAM;AACjB,aAAK;AAAA,MACP;AAAA,IACF,CAAC;AAAA,EACH;AAEA,iBAAe,iBAAgC;AAC7C,QAAI,OAAO,WAAW,aAAa;AACjC,YAAM,IAAI,MAAM,qCAAqC;AAAA,IACvD;AAEA,QAAI,CAAC,MAAM,GAAG,eAAe,UAAU,MAAM;AAC3C,YAAM,IAAI,MAAM,yBAAyB;AAAA,IAC3C;AAEA,QAAI;AACF,oBAAc,MAAM,UAAU,aAAa,aAAa;AAAA,QACtD,OAAO;AAAA,UACL;AAAA,UACA,cAAc;AAAA,UACd,kBAAkB;AAAA,UAClB,kBAAkB;AAAA,QACpB;AAAA,MACF,CAAC;AAED,qBAAe,IAAI,aAAa,EAAE,WAAW,CAAC;AAC9C,YAAM,SAAS,aAAa,wBAAwB,WAAW;AAC/D,kBAAY,aAAa,sBAAsB,MAAM,GAAG,CAAC;AAEzD,gBAAU,iBAAiB,CAAC,MAAM;AAChC,YAAI,CAAC,MAAM,GAAG,eAAe,UAAU,KAAM;AAE7C,cAAM,YAAY,EAAE,YAAY,eAAe,CAAC;AAEhD,cAAM,MAAM,IAAI,WAAW,UAAU,MAAM;AAC3C,iBAAS,IAAI,GAAG,IAAI,UAAU,QAAQ,KAAK;AACzC,gBAAM,IAAI,KAAK,IAAI,IAAI,KAAK,IAAI,GAAG,UAAU,CAAC,CAAC,CAAC;AAChD,cAAI,CAAC,IAAI,IAAI,IAAI,IAAI,QAAQ,IAAI;AAAA,QACnC;AAEA,cAAM,QAAQ,IAAI,WAAW,IAAI,MAAM;AACvC,YAAI,SAAS;AACb,iBAAS,IAAI,GAAG,IAAI,MAAM,QAAQ,KAAK;AACrC,oBAAU,OAAO,aAAa,MAAM,CAAC,CAAC;AAAA,QACxC;AACA,cAAM,SAAS,KAAK,MAAM;AAE1B,WAAG,KAAK,KAAK,UAAU;AAAA,UACrB,MAAM;AAAA,UACN,OAAO;AAAA,QACT,CAAC,CAAC;AAAA,MACJ;AAEA,aAAO,QAAQ,SAAS;AACxB,gBAAU,QAAQ,aAAa,WAAW;AAAA,IAC5C,SAAS,KAAK;AACZ,yCAAU;AACV,YAAM;AAAA,IACR;AAAA,EACF;AAEA,WAAS,gBAAgB;AACvB,QAAI,aAAa;AACf,kBAAY,UAAU,EAAE,QAAQ,WAAS,MAAM,KAAK,CAAC;AACrD,oBAAc;AAAA,IAChB;AAEA,QAAI,WAAW;AACb,gBAAU,WAAW;AACrB,kBAAY;AAAA,IACd;AAEA,QAAI,cAAc;AAChB,mBAAa,MAAM;AACnB,qBAAe;AAAA,IACjB;AAEA,QAAI,MAAM,GAAG,eAAe,UAAU,MAAM;AAC1C,SAAG,KAAK,KAAK,UAAU,EAAE,MAAM,4BAA4B,CAAC,CAAC;AAAA,IAC/D;AAAA,EACF;AAEA,WAAS,QAAQ;AACf,kBAAc;AACd,QAAI,IAAI;AACN,SAAG,MAAM;AACT,WAAK;AAAA,IACP;AAAA,EACF;AAEA,SAAO;AAAA,IACL;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,EACF;AACF;","names":[]}

package/package.json

@@ -0,0 +1,45 @@
+{
+  "name": "@amaster.ai/asr-client",
+  "version": "1.0.0-beta.1",
+  "description": "Qwen ASR Realtime WebSocket client with microphone recording",
+  "type": "module",
+  "main": "./dist/index.cjs",
+  "module": "./dist/index.js",
+  "types": "./dist/index.d.ts",
+  "exports": {
+    ".": {
+      "import": "./dist/index.js",
+      "require": "./dist/index.cjs",
+      "types": "./dist/index.d.ts"
+    }
+  },
+  "files": [
+    "dist",
+    "README.md"
+  ],
+  "keywords": [
+    "asr",
+    "speech-to-text",
+    "qwen",
+    "realtime",
+    "websocket",
+    "audio",
+    "speech-recognition"
+  ],
+  "author": "Amaster Team",
+  "license": "MIT",
+  "publishConfig": {
+    "access": "public",
+    "registry": "https://registry.npmjs.org/"
+  },
+  "devDependencies": {
+    "tsup": "^8.3.5",
+    "typescript": "~5.7.2"
+  },
+  "scripts": {
+    "build": "tsup",
+    "dev": "tsup --watch",
+    "clean": "rm -rf dist *.tsbuildinfo",
+    "type-check": "tsc --noEmit"
+  }
+}