@amaster.ai/asr-client 1.1.0-beta.49 → 1.1.0-beta.50

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Files changed (8) hide show
  1. package/README.md +200 -45
  2. package/dist/index.cjs +187 -13
  3. package/dist/index.cjs.map +1 -1
  4. package/dist/index.d.cts +56 -4
  5. package/dist/index.d.ts +56 -4
  6. package/dist/index.js +186 -12
  7. package/dist/index.js.map +1 -1
  8. package/package.json +1 -1
package/README.md CHANGED
@@ -11,11 +11,25 @@
11
11
  - 🧠 支持语音开始 / 结束事件
12
12
  - ✍️ 支持中间结果与最终转写结果
13
13
  - 🔌 简单的 WebSocket 生命周期管理
14
+ - 🌐 同时支持 WebSocket 实时识别和 HTTP 按压识别
14
15
 
15
16
  ---
16
17
 
17
18
  ## 安装与环境要求
18
19
 
20
+ ### 安装
21
+
22
+ ```bash
23
+ # npm
24
+ npm install @amaster.ai/asr-client
25
+
26
+ # pnpm
27
+ pnpm add @amaster.ai/asr-client
28
+
29
+ # yarn
30
+ yarn add @amaster.ai/asr-client
31
+ ```
32
+
19
33
  ### 浏览器要求
20
34
 
21
35
  - 支持 `getUserMedia`
@@ -28,10 +42,14 @@
28
42
 
29
43
  ## 快速开始
30
44
 
31
- ### 1️⃣ 创建 ASR Client
45
+ ### 方式一：WebSocket 实时 ASR（流式识别）
46
+
47
+ 适合需要实时看到识别结果的场景，如语音输入、实时字幕等。
48
+
49
+ #### 1️⃣ 创建 ASR Client
32
50
 
33
51
  ```ts
34
- import { createASRClient } from "./asr-client";
52
+ import { createASRClient } from "@amaster.ai/asr-client";
35
53
 
36
54
  const client = createASRClient({
37
55
  onReady() {
@@ -60,9 +78,7 @@ const client = createASRClient({
60
78
  });
61
79
  ```
62
80
 
63
- ---
64
-
65
- ### 2️⃣ 建立 WebSocket 连接
81
+ #### 2️⃣ 建立 WebSocket 连接
66
82
 
67
83
  ```ts
68
84
  await client.connect();
@@ -70,33 +86,27 @@ await client.connect();
70
86
 
71
87
  连接成功后会触发 `onReady` 回调。
72
88
 
73
- ---
74
-
75
- ### 3️⃣ 开始录音并实时识别
89
+ #### 3️⃣ 开始录音并实时识别
76
90
 
77
91
  ```ts
78
92
  await client.startRecording();
79
93
  ```
80
94
 
81
- * 自动请求麦克风权限
82
- * 自动开始推送音频流
83
- * 服务端会持续返回实时转写结果
95
+ - 自动请求麦克风权限
96
+ - 自动开始推送音频流
97
+ - 服务端会持续返回实时转写结果
84
98
 
85
- ---
86
-
87
- ### 4️⃣ 停止录音
99
+ #### 4️⃣ 停止录音
88
100
 
89
101
  ```ts
90
102
  client.stopRecording();
91
103
  ```
92
104
 
93
- * 停止麦克风采集
94
- * 向服务端发送 `input_audio_buffer.commit`
95
- * 触发最终转写结果
96
-
97
- ---
105
+ - 停止麦克风采集
106
+ - 向服务端发送 `input_audio_buffer.commit`
107
+ - 触发最终转写结果
98
108
 
99
- ### 5️⃣ 关闭连接
109
+ #### 5️⃣ 关闭连接
100
110
 
101
111
  ```ts
102
112
  client.close();
@@ -104,67 +114,212 @@ client.close();
104
114
 
105
115
  ---
106
116
 
117
+ ### 方式二：HTTP 按压式 ASR（一句话识别）
118
+
119
+ 适合按住说话、松开识别的场景，如语音消息、语音搜索等。
120
+
121
+ ```tsx
122
+ import { useRef, useState } from "react";
123
+ import { createASRHttpClient } from "@amaster.ai/asr-client";
124
+
125
+ function VoiceButton() {
126
+ const [recording, setRecording] = useState(false);
127
+ const [text, setText] = useState("");
128
+ const [error, setError] = useState<string | null>(null);
129
+
130
+ // 创建 HTTP ASR 客户端
131
+ const asrHttpClient = useRef(
132
+ createASRHttpClient({
133
+ onRecordingStart() {
134
+ setRecording(true);
135
+ setText("");
136
+ setError(null);
137
+ },
138
+ onRecordingStop() {
139
+ setRecording(false);
140
+ },
141
+ onResult(result) {
142
+ setText(result);
143
+ },
144
+ onError(err) {
145
+ setError(err.message);
146
+ },
147
+ })
148
+ ).current;
149
+
150
+ return (
151
+ <div>
152
+ <button
153
+ onMouseDown={() => asrHttpClient.startRecording()}
154
+ onMouseUp={() => asrHttpClient.stopRecording()}
155
+ onTouchStart={() => asrHttpClient.startRecording()}
156
+ onTouchEnd={() => asrHttpClient.stopRecording()}
157
+ style={{
158
+ padding: "12px 24px",
159
+ background: recording ? "#f87171" : "#4ade80",
160
+ }}
161
+ >
162
+ {recording ? "松开识别" : "按住说话"}
163
+ </button>
164
+
165
+ <div>
166
+ <strong>识别结果：</strong>
167
+ <div>{text || "（暂无）"}</div>
168
+ </div>
169
+
170
+ {error && <div style={{ color: "red" }}>错误：{error}</div>}
171
+ </div>
172
+ );
173
+ }
174
+ ```
175
+
176
+ ---
177
+
107
178
  ## API 说明
108
179
 
109
180
  ### `createASRClient(config): ASRClient`
110
181
 
182
+ WebSocket 实时 ASR 客户端。
183
+
111
184
  #### `ASRClientConfig`
112
185
 
113
- | 参数 | 类型 | 说明 |
114
- | --------------- | ------------------------------------------ | -------------- |
115
- | `onReady` | `() => void` | 会话创建完成 |
116
- | `onSpeechStart` | `() => void` | 检测到语音开始 |
117
- | `onSpeechEnd` | `() => void` | 检测到语音结束 |
118
- | `onTranscript` | `(text: string, isFinal: boolean) => void` | 转写回调 |
119
- | `onError` | `(error: Error) => void` | 错误回调 |
120
- | `onClose` | `() => void` | 连接关闭回调 |
186
+ | 参数 | 类型 | 说明 |
187
+ | ---------------- | ------------------------------------------ | -------------------------------------- |
188
+ | `audioFormat` | `"pcm16" \| "g711a" \| "g711u"` | 音频格式，默认 `'pcm16'` |
189
+ | `sampleRate` | `number` | 采样率，默认 `16000` |
190
+ | `getAccessToken` | `() => string \| null` | 获取访问令牌（用于 WebSocket 认证） |
191
+ | `onReady` | `() => void` | 会话创建完成 |
192
+ | `onSpeechStart` | `() => void` | 检测到语音开始 |
193
+ | `onSpeechEnd` | `() => void` | 检测到语音结束 |
194
+ | `onTranscript` | `(text: string, isFinal: boolean) => void` | 转写回调，`isFinal` 表示是否为最终结果 |
195
+ | `onError` | `(error: Error) => void` | 错误回调 |
196
+ | `onClose` | `() => void` | 连接关闭回调 |
197
+
198
+ #### `ASRClient`
199
+
200
+ ```ts
201
+ interface ASRClient {
202
+ connect(): Promise<void>; // 建立 WebSocket 连接
203
+ startRecording(): Promise<void>; // 开始录音
204
+ stopRecording(): void; // 停止录音
205
+ close(): void; // 关闭连接
206
+ }
207
+ ```
121
208
 
122
209
  ---
123
210
 
124
- ### `ASRClient`
211
+ ### `createASRHttpClient(config): ASRHttpClient`
212
+
213
+ HTTP 按压式 ASR 客户端。
214
+
215
+ #### `ASRHttpClientConfig`
216
+
217
+ | 参数 | 类型 | 说明 |
218
+ | ------------------ | ------------------------ | -------------------- |
219
+ | `getAccessToken` | `() => string \| null` | 获取访问令牌 |
220
+ | `language` | `string` | 语言，默认 `'zh'` |
221
+ | `sampleRate` | `number` | 采样率，默认 `16000` |
222
+ | `onRecordingStart` | `() => void` | 录音开始回调 |
223
+ | `onRecordingStop` | `() => void` | 录音停止回调 |
224
+ | `onResult` | `(text: string) => void` | 识别结果回调 |
225
+ | `onError` | `(error: Error) => void` | 错误回调 |
226
+
227
+ #### `ASRHttpClient`
125
228
 
126
229
  ```ts
127
- interface ASRClient {
128
- connect(): Promise<void>;
129
- startRecording(): Promise<void>;
130
- stopRecording(): void;
131
- close(): void;
230
+ interface ASRHttpClient {
231
+ startRecording(): Promise<void>; // 开始录音
232
+ stopRecording(): Promise<string>; // 停止录音并返回识别结果
233
+ recordAndRecognize(durationMs: number): Promise<string>; // 录音指定时长
234
+ recognizeFile(file: File | Blob): Promise<string>; // 识别音频文件
235
+ recognizeUrl(audioUrl: string): Promise<string>; // 识别音频 URL
132
236
  }
133
237
  ```
134
238
 
135
239
  ---
136
240
 
241
+ ## 配合统一客户端使用
242
+
243
+ 推荐与 `@amaster.ai/client` 统一客户端一起使用，自动处理认证：
244
+
245
+ ```tsx
246
+ import { createClient } from "@amaster.ai/client";
247
+
248
+ const client = createClient({
249
+ baseURL: "https://api.amaster.ai",
250
+ });
251
+
252
+ // WebSocket ASR
253
+ const asrClient = client.asr({
254
+ onTranscript(text, isFinal) {
255
+ console.log(isFinal ? "[最终]" : "[实时]", text);
256
+ },
257
+ });
258
+
259
+ await asrClient.connect();
260
+ await asrClient.startRecording();
261
+
262
+ // HTTP ASR
263
+ const asrHttpClient = client.asrHttp({
264
+ onResult(text) {
265
+ console.log("识别结果:", text);
266
+ },
267
+ });
268
+
269
+ await asrHttpClient.startRecording();
270
+ // ... 用户说话 ...
271
+ const result = await asrHttpClient.stopRecording();
272
+ ```
273
+
274
+ ---
275
+
137
276
  ## 音频参数说明
138
277
 
139
- * **采样率**：16000 Hz
140
- * **声道**：单声道
141
- * **格式**：PCM16 → Base64
142
- * **缓冲大小**：4096 frames
278
+ ### WebSocket ASR
279
+
280
+ - **采样率**：16000 Hz
281
+ - **声道**：单声道
282
+ - **格式**：PCM16 → Base64
283
+ - **缓冲大小**：4096 frames
284
+
285
+ ### HTTP ASR
286
+
287
+ - **采样率**：自动检测（通常为 16000 Hz）
288
+ - **声道**：单声道
289
+ - **格式**：WAV（内部将 PCM 转为 WAV）
143
290
 
144
291
  ---
145
292
 
146
293
  ## 常见问题
147
294
 
295
+ ### Q: WebSocket 和 HTTP ASR 有什么区别？
296
+
297
+ | 特性 | WebSocket ASR | HTTP ASR |
298
+ | -------- | ---------------------- | ------------------- |
299
+ | 实时性 | 实时流式返回 | 说完后一次性返回 |
300
+ | 适用场景 | 语音输入、实时字幕 | 语音消息、语音搜索 |
301
+ | 交互方式 | 开始 → 持续识别 → 停止 | 按住说话 → 松开识别 |
302
+ | 网络开销 | 持续连接 | 每次请求独立 |
303
+
148
304
  ### Q: 为什么必须是 16kHz？
149
305
 
150
306
  ASR 服务通常要求 16kHz PCM 输入，否则会影响识别效果或直接报错。
151
307
 
152
- ---
153
-
154
308
  ### Q: 支持移动端吗？
155
309
 
156
310
  支持，但需注意：
157
311
 
158
- * iOS Safari 需用户手势触发录音
159
- * 后台会自动暂停音频采集
312
+ - iOS Safari 需用户手势触发录音
313
+ - 后台会自动暂停音频采集
160
314
 
161
315
  ---
162
316
 
163
317
  ## 注意事项
164
318
 
165
- * WebSocket 必须在 HTTPS 页面下使用麦克风
166
- * 页面关闭前建议调用 `client.close()`
167
- * 不建议在多个 ASR Client 实例中共享麦克风
319
+ - WebSocket 必须在 HTTPS 页面下使用麦克风
320
+ - 页面关闭前建议调用 `client.close()`
321
+ - 不建议在多个 ASR Client 实例中共享麦克风
322
+ - HTTP ASR 会自动将录音转为 WAV 格式上传
168
323
 
169
324
  ---
170
325
 
package/dist/index.cjs CHANGED
@@ -20,8 +20,8 @@ var __toCommonJS = (mod) => __copyProps(__defProp({}, "__esModule", { value: tru
20
20
  // src/index.ts
21
21
  var index_exports = {};
22
22
  __export(index_exports, {
23
- createASRClient: () => createASRClient,
24
- createAsrClient: () => createASRClient
23
+ createASRClient: () => asr_client_default,
24
+ createASRHttpClient: () => http_asr_client_default
25
25
  });
26
26
  module.exports = __toCommonJS(index_exports);
27
27
 
@@ -71,18 +71,10 @@ var log = (message, type = "") => {
71
71
  console.log(`[${type}]`, message);
72
72
  };
73
73
  function createASRClient(config) {
74
- const {
75
- onReady,
76
- onSpeechStart,
77
- onSpeechEnd,
78
- onTranscript,
79
- onError,
80
- onClose,
81
- path = ASR_PATH,
82
- getAccessToken
83
- } = config;
74
+ const { onReady, onSpeechStart, onSpeechEnd, onTranscript, onError, onClose, getAccessToken } = config;
84
75
  let ws = null;
85
76
  let recorder = null;
77
+ const path = ASR_PATH;
86
78
  async function connect() {
87
79
  let wsUrl = path;
88
80
  if (getAccessToken) {
@@ -168,9 +160,191 @@ function createASRClient(config) {
168
160
  close
169
161
  };
170
162
  }
163
+ var asr_client_default = (authConfig) => (config) => createASRClient({ ...authConfig, ...config });
164
+
165
+ // src/http-asr-client.ts
166
+ var ASR_HTTP_PATH = "/api/proxy/builtin/platform/qwen-asr/compatible-mode/v1/chat/completions";
167
+ var RECORDER_WORKLET = `
168
+ class RecorderProcessor extends AudioWorkletProcessor {
169
+ process(inputs) {
170
+ const input = inputs[0];
171
+ if (input && input[0]) {
172
+ this.port.postMessage(input[0].slice(0));
173
+ }
174
+ return true;
175
+ }
176
+ }
177
+ registerProcessor('recorder-processor', RecorderProcessor);
178
+ `;
179
+ async function createRecorder() {
180
+ let stream;
181
+ let ctx;
182
+ let node;
183
+ let source;
184
+ const chunks = [];
185
+ return {
186
+ async start() {
187
+ stream = await navigator.mediaDevices.getUserMedia({
188
+ audio: {
189
+ channelCount: 1,
190
+ echoCancellation: true,
191
+ noiseSuppression: true,
192
+ autoGainControl: true
193
+ }
194
+ });
195
+ ctx = new AudioContext();
196
+ const blob = new Blob([RECORDER_WORKLET], { type: "application/javascript" });
197
+ const url = URL.createObjectURL(blob);
198
+ await ctx.audioWorklet.addModule(url);
199
+ URL.revokeObjectURL(url);
200
+ source = ctx.createMediaStreamSource(stream);
201
+ node = new AudioWorkletNode(ctx, "recorder-processor");
202
+ node.port.onmessage = (e) => {
203
+ const input = e.data;
204
+ const pcm = new Int16Array(input.length);
205
+ for (let i = 0; i < input.length; i++) {
206
+ const s = Math.max(-1, Math.min(1, input[i] || 0));
207
+ pcm[i] = s < 0 ? s * 32768 : s * 32767;
208
+ }
209
+ chunks.push(pcm);
210
+ };
211
+ source.connect(node);
212
+ },
213
+ async stop() {
214
+ source?.disconnect();
215
+ node?.disconnect();
216
+ stream?.getTracks().forEach((t) => t.stop());
217
+ if (ctx) {
218
+ await ctx.close();
219
+ }
220
+ const total = chunks.reduce((s, c) => s + c.length, 0);
221
+ const pcm = new Int16Array(total);
222
+ let offset = 0;
223
+ for (const c of chunks) {
224
+ pcm.set(c, offset);
225
+ offset += c.length;
226
+ }
227
+ return { pcm, sampleRate: ctx?.sampleRate ?? 16e3 };
228
+ }
229
+ };
230
+ }
231
+ function pcmToWav(pcm, sampleRate) {
232
+ const buffer = new ArrayBuffer(44 + pcm.length * 2);
233
+ const view = new DataView(buffer);
234
+ const write = (o, s) => {
235
+ for (let i = 0; i < s.length; i++) view.setUint8(o + i, s.charCodeAt(i));
236
+ };
237
+ write(0, "RIFF");
238
+ view.setUint32(4, 36 + pcm.length * 2, true);
239
+ write(8, "WAVE");
240
+ write(12, "fmt ");
241
+ view.setUint32(16, 16, true);
242
+ view.setUint16(20, 1, true);
243
+ view.setUint16(22, 1, true);
244
+ view.setUint32(24, sampleRate, true);
245
+ view.setUint32(28, sampleRate * 2, true);
246
+ view.setUint16(32, 2, true);
247
+ view.setUint16(34, 16, true);
248
+ write(36, "data");
249
+ view.setUint32(40, pcm.length * 2, true);
250
+ for (let i = 0; i < pcm.length; i++) {
251
+ view.setInt16(44 + i * 2, pcm[i] || 0, true);
252
+ }
253
+ return new Blob([buffer], { type: "audio/wav" });
254
+ }
255
+ function blobToBase64(blob) {
256
+ return new Promise((resolve, reject) => {
257
+ const reader = new FileReader();
258
+ reader.onloadend = () => {
259
+ const result = reader.result;
260
+ resolve(result.split(",")[1] || "");
261
+ };
262
+ reader.onerror = reject;
263
+ reader.readAsDataURL(blob);
264
+ });
265
+ }
266
+ async function recognizeBlob(blob, path) {
267
+ const base64 = await blobToBase64(blob);
268
+ const response = await fetch(path ?? ASR_HTTP_PATH, {
269
+ method: "POST",
270
+ headers: { "Content-Type": "application/json" },
271
+ body: JSON.stringify({
272
+ model: "qwen3-asr-flash",
273
+ messages: [
274
+ {
275
+ role: "user",
276
+ content: [
277
+ {
278
+ type: "input_audio",
279
+ input_audio: { data: `data:audio/wav;base64,${base64}` }
280
+ }
281
+ ]
282
+ }
283
+ ]
284
+ })
285
+ });
286
+ const data = await response.json();
287
+ return data.choices?.[0]?.message?.content || "";
288
+ }
289
+ async function recognizeFile(file, path) {
290
+ return recognizeBlob(file, path);
291
+ }
292
+ function createASRHttpClient(config) {
293
+ let recorder = null;
294
+ let path = ASR_HTTP_PATH;
295
+ if (config.getAccessToken) {
296
+ const token = config.getAccessToken();
297
+ if (token) {
298
+ const separator = path.includes("?") ? "&" : "?";
299
+ path = `${path}${separator}token=${encodeURIComponent(token)}`;
300
+ }
301
+ }
302
+ return {
303
+ async startRecording() {
304
+ recorder = await createRecorder();
305
+ await recorder.start();
306
+ config.onRecordingStart?.();
307
+ },
308
+ async stopRecording() {
309
+ if (!recorder) throw new Error("Not recording");
310
+ const { pcm, sampleRate } = await recorder.stop();
311
+ recorder = null;
312
+ config.onRecordingStop?.();
313
+ const text = await recognizeBlob(pcmToWav(pcm, sampleRate), path);
314
+ config.onResult?.(text);
315
+ return text;
316
+ },
317
+ async recordAndRecognize(ms) {
318
+ await this.startRecording();
319
+ await new Promise((r) => setTimeout(r, ms));
320
+ return this.stopRecording();
321
+ },
322
+ recognizeFile: (file) => {
323
+ return recognizeFile(file, path);
324
+ },
325
+ async recognizeUrl(url) {
326
+ const res = await fetch(path, {
327
+ method: "POST",
328
+ headers: { "Content-Type": "application/json" },
329
+ body: JSON.stringify({
330
+ model: "qwen3-asr-flash",
331
+ messages: [
332
+ {
333
+ role: "user",
334
+ content: [{ type: "input_audio", input_audio: { url } }]
335
+ }
336
+ ]
337
+ })
338
+ });
339
+ const data = await res.json();
340
+ return data.choices?.[0]?.message?.content || "";
341
+ }
342
+ };
343
+ }
344
+ var http_asr_client_default = (authConfig) => (config) => createASRHttpClient({ ...authConfig, ...config });
171
345
  // Annotate the CommonJS export names for ESM import in node:
172
346
  0 && (module.exports = {
173
347
  createASRClient,
174
- createAsrClient
348
+ createASRHttpClient
175
349
  });
176
350
  //# sourceMappingURL=index.cjs.map
package/dist/index.cjs.map CHANGED
@@ -1 +1 @@
1
- {"version":3,"sources":["../src/index.ts","../src/asr-client.ts"],"sourcesContent":["export * from './asr-client';\nexport { createASRClient as createAsrClient } from './asr-client';\n","/**\n * ASR Realtime WebSocket Client\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\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;\n let ctx: AudioContext;\n let source: MediaStreamAudioSourceNode;\n let processor: ScriptProcessorNode;\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 await ctx?.close();\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\nexport interface ASRClientConfig {\n /** Audio format, default 'pcm16' */\n audioFormat?: \"pcm16\" | \"g711a\" | \"g711u\";\n /** Sample rate, default 16000 */\n sampleRate?: number;\n /** Get access token for WebSocket authentication */\n getAccessToken?: () => string | null;\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 /** Called on close */\n onClose?: () => void;\n /** WebSocket path, default ASR_PATH */\n path?: string;\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 onReady,\n onSpeechStart,\n onSpeechEnd,\n onTranscript,\n onError,\n onClose,\n path = ASR_PATH,\n getAccessToken,\n } = config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\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('?') ? 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1
+ {"version":3,"sources":["../src/index.ts","../src/asr-client.ts","../src/http-asr-client.ts"],"sourcesContent":["import createASRClient, { type ASRClientConfig, type ASRClient } from \"./asr-client\";\nexport type { ASRClient, ASRClientConfig };\nexport { createASRClient };\n\nimport createASRHttpClient, { type ASRHttpClientConfig, type ASRHttpClient } from \"./http-asr-client\";\nexport type { ASRHttpClient, ASRHttpClientConfig };\nexport { createASRHttpClient };","/**\n * ASR Realtime WebSocket Client\n *\n * WebSocket-based real-time speech recognition for streaming transcription.\n * Suitable for scenarios requiring real-time results like voice input, live captions, etc.\n *\n * @example\n * ```typescript\n * const client = createASRClient({\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 * // ... speak ...\n * client.stopRecording();\n * client.close();\n * ```\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\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;\n let ctx: AudioContext;\n let source: MediaStreamAudioSourceNode;\n let processor: ScriptProcessorNode;\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 await ctx?.close();\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\nexport interface ASRClientConfig {\n /** Audio format, default 'pcm16' */\n audioFormat?: \"pcm16\" | \"g711a\" | \"g711u\";\n /** Sample rate, default 16000 */\n sampleRate?: number;\n /** Get access token for WebSocket authentication */\n getAccessToken?: () => string | null;\n /** Called when connection is ready */\n onReady?: () => void;\n /** Called when speech is detected */\n onSpeechStart?: () => void;\n /** Called when speech stops */\n onSpeechEnd?: () => void;\n /** Called on transcript result */\n onTranscript?: (text: string, isFinal: boolean) => void;\n /** Called on error */\n onError?: (error: Error) => void;\n /** Called on close */\n onClose?: () => void;\n}\n\nexport interface ASRClient {\n /** Connect to ASR service */\n connect(): Promise<void>;\n /** Start recording from microphone */\n startRecording(): Promise<void>;\n /** Stop recording */\n stopRecording(): void;\n /** Close connection */\n close(): void;\n}\n\nfunction createASRClient(config: ASRClientConfig): ASRClient {\n const { onReady, onSpeechStart, onSpeechEnd, onTranscript, onError, onClose, getAccessToken } =\n config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\n\n const path = ASR_PATH;\n\n async function connect(): Promise<void> {\n // Build WebSocket URL with optional token parameter\n let wsUrl = path;\n if (getAccessToken) {\n const token = getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n wsUrl = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n ws = new WebSocket(wsUrl);\n\n return new Promise((resolve, reject) => {\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 || data.stash || data.transcript || \"\", 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 || \"ASR error\");\n onError?.(err);\n reject(err);\n }\n };\n\n ws!.onerror = () => {\n const err = new Error(\"WebSocket error\");\n onError?.(err);\n reject(err);\n };\n\n ws!.onclose = () => {\n recorder?.stop();\n recorder = null;\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 recorder = await createRealtimeRecorder();\n await recorder.start((audio) => {\n if (!ws || ws.readyState !== WebSocket.OPEN) return;\n ws.send(\n JSON.stringify({\n type: \"input_audio_buffer.append\",\n audio,\n })\n );\n });\n }\n\n async function stopRecording() {\n try {\n await recorder?.stop();\n } catch (err) {}\n recorder = null;\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 ws?.close();\n ws = null;\n }\n\n return {\n connect,\n startRecording,\n stopRecording,\n close,\n };\n}\n\nexport default (authConfig: Pick<ASRClientConfig, \"getAccessToken\">) => (config: ASRClientConfig) =>\n createASRClient({ ...authConfig, ...config });\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 * // ... speak ...\n * const result = await client.stopRecording();\n * console.log(\"Result:\", result);\n * ```\n */\n\nconst ASR_HTTP_PATH = \"/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\ninterface Recorder {\n start(): Promise<void>;\n stop(): Promise<{ pcm: Int16Array; sampleRate: number }>;\n}\n\n// 创建录音器\nasync function createRecorder(): Promise<Recorder> {\n let stream: MediaStream;\n let ctx: AudioContext;\n let node: AudioWorkletNode;\n let source: MediaStreamAudioSourceNode;\n const chunks: Int16Array[] = [];\n\n return {\n async start() {\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], { type: \"application/javascript\" });\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\n async stop() {\n source?.disconnect();\n node?.disconnect();\n stream?.getTracks().forEach((t) => t.stop());\n if (ctx) {\n await ctx.close();\n }\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 return { pcm, sampleRate: ctx?.sampleRate ?? 16000 };\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 return new Blob([buffer], { type: \"audio/wav\" });\n}\n\nfunction blobToBase64(blob: Blob): Promise<string> {\n return new Promise((resolve, reject) => {\n const reader = new FileReader();\n reader.onloadend = () => {\n const result = reader.result as string;\n resolve(result.split(\",\")[1] || \"\");\n };\n reader.onerror = reject;\n reader.readAsDataURL(blob);\n });\n}\n\n/**\n * Recognize from Blob\n * @param blob - Audio Blob\n * @returns Recognized text\n */\nasync function recognizeBlob(blob: Blob, path?: string): Promise<string> {\n const base64 = await blobToBase64(blob);\n const response = await fetch(path ?? ASR_HTTP_PATH, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: 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,${base64}` },\n },\n ],\n },\n ],\n }),\n });\n\n const data = await response.json();\n return data.choices?.[0]?.message?.content || \"\";\n}\n\n/** Recognize audio file */\nexport async function recognizeFile(file: File | Blob, path?: string): Promise<string> {\n return recognizeBlob(file, path);\n}\n\nexport interface ASRHttpClientConfig {\n /** Get access token */\n getAccessToken?(): string | null;\n /** Language, default 'zh' */\n language?: string;\n /** Sample rate, default 16000 */\n sampleRate?: number;\n /** Called when recording starts */\n onRecordingStart?: () => void;\n /** Called when recording stops */\n onRecordingStop?: () => void;\n /** Called with recognition result */\n onResult?: (text: string) => void;\n /** Called on error */\n onError?: (error: Error) => void;\n}\n\nexport interface ASRHttpClient {\n /** Start recording (press-to-talk) */\n startRecording(): Promise<void>;\n /** Stop recording and get result */\n stopRecording(): Promise<string>;\n /** Record for specific duration then recognize */\n recordAndRecognize(durationMs: number): Promise<string>;\n /** Recognize audio file (File or Blob) */\n recognizeFile(file: File | Blob): Promise<string>;\n /** Recognize audio from URL */\n recognizeUrl(audioUrl: string): Promise<string>;\n}\n\nexport function createASRHttpClient(config: ASRHttpClientConfig): ASRHttpClient {\n let recorder: Recorder | null = null;\n\n let path = ASR_HTTP_PATH;\n\n if (config.getAccessToken) {\n const token = config.getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n path = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n return {\n async startRecording() {\n recorder = await createRecorder();\n await recorder.start();\n config.onRecordingStart?.();\n },\n\n async stopRecording() {\n if (!recorder) throw new Error(\"Not recording\");\n const { pcm, sampleRate } = await recorder.stop();\n recorder = null;\n config.onRecordingStop?.();\n const text = await recognizeBlob(pcmToWav(pcm, sampleRate), path);\n config.onResult?.(text);\n return text;\n },\n\n async recordAndRecognize(ms: number) {\n await this.startRecording();\n await new Promise((r) => setTimeout(r, ms));\n return this.stopRecording();\n },\n\n recognizeFile: (file: File | Blob) => {\n return recognizeFile(file, path);\n },\n\n async recognizeUrl(url: string) {\n const res = await fetch(path, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: JSON.stringify({\n model: \"qwen3-asr-flash\",\n messages: [\n {\n role: \"user\",\n content: [{ type: \"input_audio\", input_audio: { url } }],\n },\n ],\n }),\n });\n const data = await res.json();\n return 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package/dist/index.d.cts CHANGED
@@ -1,5 +1,29 @@
1
1
  /**
2
2
  * ASR Realtime WebSocket Client
3
+ *
4
+ * WebSocket-based real-time speech recognition for streaming transcription.
5
+ * Suitable for scenarios requiring real-time results like voice input, live captions, etc.
6
+ *
7
+ * @example
8
+ * ```typescript
9
+ * const client = createASRClient({
10
+ * onReady() {
11
+ * console.log("ASR connected");
12
+ * },
13
+ * onTranscript(text, isFinal) {
14
+ * console.log(isFinal ? "[Final]" : "[Interim]", text);
15
+ * },
16
+ * onError(err) {
17
+ * console.error("ASR error:", err);
18
+ * },
19
+ * });
20
+ *
21
+ * await client.connect();
22
+ * await client.startRecording();
23
+ * // ... speak ...
24
+ * client.stopRecording();
25
+ * client.close();
26
+ * ```
3
27
  */
4
28
  interface ASRClientConfig {
5
29
  /** Audio format, default 'pcm16' */
@@ -20,8 +44,6 @@ interface ASRClientConfig {
20
44
  onError?: (error: Error) => void;
21
45
  /** Called on close */
22
46
  onClose?: () => void;
23
- /** WebSocket path, default ASR_PATH */
24
- path?: string;
25
47
  }
26
48
  interface ASRClient {
27
49
  /** Connect to ASR service */
@@ -33,6 +55,36 @@ interface ASRClient {
33
55
  /** Close connection */
34
56
  close(): void;
35
57
  }
36
- declare function createASRClient(config: ASRClientConfig): ASRClient;
58
+ declare const _default$1: (authConfig: Pick<ASRClientConfig, "getAccessToken">) => (config: ASRClientConfig) => ASRClient;
37
59
 
38
- export { type ASRClient, type ASRClientConfig, createASRClient, createASRClient as createAsrClient };
60
+ interface ASRHttpClientConfig {
61
+ /** Get access token */
62
+ getAccessToken?(): string | null;
63
+ /** Language, default 'zh' */
64
+ language?: string;
65
+ /** Sample rate, default 16000 */
66
+ sampleRate?: number;
67
+ /** Called when recording starts */
68
+ onRecordingStart?: () => void;
69
+ /** Called when recording stops */
70
+ onRecordingStop?: () => void;
71
+ /** Called with recognition result */
72
+ onResult?: (text: string) => void;
73
+ /** Called on error */
74
+ onError?: (error: Error) => void;
75
+ }
76
+ interface ASRHttpClient {
77
+ /** Start recording (press-to-talk) */
78
+ startRecording(): Promise<void>;
79
+ /** Stop recording and get result */
80
+ stopRecording(): Promise<string>;
81
+ /** Record for specific duration then recognize */
82
+ recordAndRecognize(durationMs: number): Promise<string>;
83
+ /** Recognize audio file (File or Blob) */
84
+ recognizeFile(file: File | Blob): Promise<string>;
85
+ /** Recognize audio from URL */
86
+ recognizeUrl(audioUrl: string): Promise<string>;
87
+ }
88
+ declare const _default: (authConfig: Pick<ASRHttpClientConfig, "getAccessToken">) => (config: ASRHttpClientConfig) => ASRHttpClient;
89
+
90
+ export { type ASRClient, type ASRClientConfig, type ASRHttpClient, type ASRHttpClientConfig, _default$1 as createASRClient, _default as createASRHttpClient };
package/dist/index.d.ts CHANGED
@@ -1,5 +1,29 @@
1
1
  /**
2
2
  * ASR Realtime WebSocket Client
3
+ *
4
+ * WebSocket-based real-time speech recognition for streaming transcription.
5
+ * Suitable for scenarios requiring real-time results like voice input, live captions, etc.
6
+ *
7
+ * @example
8
+ * ```typescript
9
+ * const client = createASRClient({
10
+ * onReady() {
11
+ * console.log("ASR connected");
12
+ * },
13
+ * onTranscript(text, isFinal) {
14
+ * console.log(isFinal ? "[Final]" : "[Interim]", text);
15
+ * },
16
+ * onError(err) {
17
+ * console.error("ASR error:", err);
18
+ * },
19
+ * });
20
+ *
21
+ * await client.connect();
22
+ * await client.startRecording();
23
+ * // ... speak ...
24
+ * client.stopRecording();
25
+ * client.close();
26
+ * ```
3
27
  */
4
28
  interface ASRClientConfig {
5
29
  /** Audio format, default 'pcm16' */
@@ -20,8 +44,6 @@ interface ASRClientConfig {
20
44
  onError?: (error: Error) => void;
21
45
  /** Called on close */
22
46
  onClose?: () => void;
23
- /** WebSocket path, default ASR_PATH */
24
- path?: string;
25
47
  }
26
48
  interface ASRClient {
27
49
  /** Connect to ASR service */
@@ -33,6 +55,36 @@ interface ASRClient {
33
55
  /** Close connection */
34
56
  close(): void;
35
57
  }
36
- declare function createASRClient(config: ASRClientConfig): ASRClient;
58
+ declare const _default$1: (authConfig: Pick<ASRClientConfig, "getAccessToken">) => (config: ASRClientConfig) => ASRClient;
37
59
 
38
- export { type ASRClient, type ASRClientConfig, createASRClient, createASRClient as createAsrClient };
60
+ interface ASRHttpClientConfig {
61
+ /** Get access token */
62
+ getAccessToken?(): string | null;
63
+ /** Language, default 'zh' */
64
+ language?: string;
65
+ /** Sample rate, default 16000 */
66
+ sampleRate?: number;
67
+ /** Called when recording starts */
68
+ onRecordingStart?: () => void;
69
+ /** Called when recording stops */
70
+ onRecordingStop?: () => void;
71
+ /** Called with recognition result */
72
+ onResult?: (text: string) => void;
73
+ /** Called on error */
74
+ onError?: (error: Error) => void;
75
+ }
76
+ interface ASRHttpClient {
77
+ /** Start recording (press-to-talk) */
78
+ startRecording(): Promise<void>;
79
+ /** Stop recording and get result */
80
+ stopRecording(): Promise<string>;
81
+ /** Record for specific duration then recognize */
82
+ recordAndRecognize(durationMs: number): Promise<string>;
83
+ /** Recognize audio file (File or Blob) */
84
+ recognizeFile(file: File | Blob): Promise<string>;
85
+ /** Recognize audio from URL */
86
+ recognizeUrl(audioUrl: string): Promise<string>;
87
+ }
88
+ declare const _default: (authConfig: Pick<ASRHttpClientConfig, "getAccessToken">) => (config: ASRHttpClientConfig) => ASRHttpClient;
89
+
90
+ export { type ASRClient, type ASRClientConfig, type ASRHttpClient, type ASRHttpClientConfig, _default$1 as createASRClient, _default as createASRHttpClient };
package/dist/index.js CHANGED
@@ -44,18 +44,10 @@ var log = (message, type = "") => {
44
44
  console.log(`[${type}]`, message);
45
45
  };
46
46
  function createASRClient(config) {
47
- const {
48
- onReady,
49
- onSpeechStart,
50
- onSpeechEnd,
51
- onTranscript,
52
- onError,
53
- onClose,
54
- path = ASR_PATH,
55
- getAccessToken
56
- } = config;
47
+ const { onReady, onSpeechStart, onSpeechEnd, onTranscript, onError, onClose, getAccessToken } = config;
57
48
  let ws = null;
58
49
  let recorder = null;
50
+ const path = ASR_PATH;
59
51
  async function connect() {
60
52
  let wsUrl = path;
61
53
  if (getAccessToken) {
@@ -141,8 +133,190 @@ function createASRClient(config) {
141
133
  close
142
134
  };
143
135
  }
136
+ var asr_client_default = (authConfig) => (config) => createASRClient({ ...authConfig, ...config });
137
+
138
+ // src/http-asr-client.ts
139
+ var ASR_HTTP_PATH = "/api/proxy/builtin/platform/qwen-asr/compatible-mode/v1/chat/completions";
140
+ var RECORDER_WORKLET = `
141
+ class RecorderProcessor extends AudioWorkletProcessor {
142
+ process(inputs) {
143
+ const input = inputs[0];
144
+ if (input && input[0]) {
145
+ this.port.postMessage(input[0].slice(0));
146
+ }
147
+ return true;
148
+ }
149
+ }
150
+ registerProcessor('recorder-processor', RecorderProcessor);
151
+ `;
152
+ async function createRecorder() {
153
+ let stream;
154
+ let ctx;
155
+ let node;
156
+ let source;
157
+ const chunks = [];
158
+ return {
159
+ async start() {
160
+ stream = await navigator.mediaDevices.getUserMedia({
161
+ audio: {
162
+ channelCount: 1,
163
+ echoCancellation: true,
164
+ noiseSuppression: true,
165
+ autoGainControl: true
166
+ }
167
+ });
168
+ ctx = new AudioContext();
169
+ const blob = new Blob([RECORDER_WORKLET], { type: "application/javascript" });
170
+ const url = URL.createObjectURL(blob);
171
+ await ctx.audioWorklet.addModule(url);
172
+ URL.revokeObjectURL(url);
173
+ source = ctx.createMediaStreamSource(stream);
174
+ node = new AudioWorkletNode(ctx, "recorder-processor");
175
+ node.port.onmessage = (e) => {
176
+ const input = e.data;
177
+ const pcm = new Int16Array(input.length);
178
+ for (let i = 0; i < input.length; i++) {
179
+ const s = Math.max(-1, Math.min(1, input[i] || 0));
180
+ pcm[i] = s < 0 ? s * 32768 : s * 32767;
181
+ }
182
+ chunks.push(pcm);
183
+ };
184
+ source.connect(node);
185
+ },
186
+ async stop() {
187
+ source?.disconnect();
188
+ node?.disconnect();
189
+ stream?.getTracks().forEach((t) => t.stop());
190
+ if (ctx) {
191
+ await ctx.close();
192
+ }
193
+ const total = chunks.reduce((s, c) => s + c.length, 0);
194
+ const pcm = new Int16Array(total);
195
+ let offset = 0;
196
+ for (const c of chunks) {
197
+ pcm.set(c, offset);
198
+ offset += c.length;
199
+ }
200
+ return { pcm, sampleRate: ctx?.sampleRate ?? 16e3 };
201
+ }
202
+ };
203
+ }
204
+ function pcmToWav(pcm, sampleRate) {
205
+ const buffer = new ArrayBuffer(44 + pcm.length * 2);
206
+ const view = new DataView(buffer);
207
+ const write = (o, s) => {
208
+ for (let i = 0; i < s.length; i++) view.setUint8(o + i, s.charCodeAt(i));
209
+ };
210
+ write(0, "RIFF");
211
+ view.setUint32(4, 36 + pcm.length * 2, true);
212
+ write(8, "WAVE");
213
+ write(12, "fmt ");
214
+ view.setUint32(16, 16, true);
215
+ view.setUint16(20, 1, true);
216
+ view.setUint16(22, 1, true);
217
+ view.setUint32(24, sampleRate, true);
218
+ view.setUint32(28, sampleRate * 2, true);
219
+ view.setUint16(32, 2, true);
220
+ view.setUint16(34, 16, true);
221
+ write(36, "data");
222
+ view.setUint32(40, pcm.length * 2, true);
223
+ for (let i = 0; i < pcm.length; i++) {
224
+ view.setInt16(44 + i * 2, pcm[i] || 0, true);
225
+ }
226
+ return new Blob([buffer], { type: "audio/wav" });
227
+ }
228
+ function blobToBase64(blob) {
229
+ return new Promise((resolve, reject) => {
230
+ const reader = new FileReader();
231
+ reader.onloadend = () => {
232
+ const result = reader.result;
233
+ resolve(result.split(",")[1] || "");
234
+ };
235
+ reader.onerror = reject;
236
+ reader.readAsDataURL(blob);
237
+ });
238
+ }
239
+ async function recognizeBlob(blob, path) {
240
+ const base64 = await blobToBase64(blob);
241
+ const response = await fetch(path ?? ASR_HTTP_PATH, {
242
+ method: "POST",
243
+ headers: { "Content-Type": "application/json" },
244
+ body: JSON.stringify({
245
+ model: "qwen3-asr-flash",
246
+ messages: [
247
+ {
248
+ role: "user",
249
+ content: [
250
+ {
251
+ type: "input_audio",
252
+ input_audio: { data: `data:audio/wav;base64,${base64}` }
253
+ }
254
+ ]
255
+ }
256
+ ]
257
+ })
258
+ });
259
+ const data = await response.json();
260
+ return data.choices?.[0]?.message?.content || "";
261
+ }
262
+ async function recognizeFile(file, path) {
263
+ return recognizeBlob(file, path);
264
+ }
265
+ function createASRHttpClient(config) {
266
+ let recorder = null;
267
+ let path = ASR_HTTP_PATH;
268
+ if (config.getAccessToken) {
269
+ const token = config.getAccessToken();
270
+ if (token) {
271
+ const separator = path.includes("?") ? "&" : "?";
272
+ path = `${path}${separator}token=${encodeURIComponent(token)}`;
273
+ }
274
+ }
275
+ return {
276
+ async startRecording() {
277
+ recorder = await createRecorder();
278
+ await recorder.start();
279
+ config.onRecordingStart?.();
280
+ },
281
+ async stopRecording() {
282
+ if (!recorder) throw new Error("Not recording");
283
+ const { pcm, sampleRate } = await recorder.stop();
284
+ recorder = null;
285
+ config.onRecordingStop?.();
286
+ const text = await recognizeBlob(pcmToWav(pcm, sampleRate), path);
287
+ config.onResult?.(text);
288
+ return text;
289
+ },
290
+ async recordAndRecognize(ms) {
291
+ await this.startRecording();
292
+ await new Promise((r) => setTimeout(r, ms));
293
+ return this.stopRecording();
294
+ },
295
+ recognizeFile: (file) => {
296
+ return recognizeFile(file, path);
297
+ },
298
+ async recognizeUrl(url) {
299
+ const res = await fetch(path, {
300
+ method: "POST",
301
+ headers: { "Content-Type": "application/json" },
302
+ body: JSON.stringify({
303
+ model: "qwen3-asr-flash",
304
+ messages: [
305
+ {
306
+ role: "user",
307
+ content: [{ type: "input_audio", input_audio: { url } }]
308
+ }
309
+ ]
310
+ })
311
+ });
312
+ const data = await res.json();
313
+ return data.choices?.[0]?.message?.content || "";
314
+ }
315
+ };
316
+ }
317
+ var http_asr_client_default = (authConfig) => (config) => createASRHttpClient({ ...authConfig, ...config });
144
318
  export {
145
- createASRClient,
146
- createASRClient as createAsrClient
319
+ asr_client_default as createASRClient,
320
+ http_asr_client_default as createASRHttpClient
147
321
  };
148
322
  //# sourceMappingURL=index.js.map
package/dist/index.js.map CHANGED
@@ -1 +1 @@
1
- {"version":3,"sources":["../src/asr-client.ts"],"sourcesContent":["/**\n * ASR Realtime WebSocket Client\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\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;\n let ctx: AudioContext;\n let source: MediaStreamAudioSourceNode;\n let processor: ScriptProcessorNode;\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 await ctx?.close();\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\nexport interface ASRClientConfig {\n /** Audio format, default 'pcm16' */\n audioFormat?: \"pcm16\" | \"g711a\" | \"g711u\";\n /** Sample rate, default 16000 */\n sampleRate?: number;\n /** Get access token for WebSocket authentication */\n getAccessToken?: () => string | null;\n /** Called when connection is ready */\n onReady?: () => void;\n /** Called when speech is detected */\n onSpeechStart?: () => void;\n /** Called when speech stops */\n onSpeechEnd?: () => void;\n /** Called on transcript result */\n onTranscript?: (text: string, isFinal: boolean) => void;\n /** Called on error */\n onError?: (error: Error) => void;\n /** Called on close */\n onClose?: () => void;\n /** WebSocket path, default ASR_PATH */\n path?: string;\n}\n\nexport interface ASRClient {\n /** Connect to ASR service */\n connect(): Promise<void>;\n /** Start recording from microphone */\n startRecording(): Promise<void>;\n /** Stop recording */\n stopRecording(): void;\n /** Close connection */\n close(): void;\n}\n\nexport function createASRClient(config: ASRClientConfig): ASRClient {\n const {\n onReady,\n onSpeechStart,\n onSpeechEnd,\n onTranscript,\n onError,\n onClose,\n path = ASR_PATH,\n getAccessToken,\n } = config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\n\n async function connect(): Promise<void> {\n // Build WebSocket URL with optional token parameter\n let wsUrl = path;\n if (getAccessToken) {\n const token = getAccessToken();\n if (token) {\n const separator = path.includes('?') ? '&' : '?';\n wsUrl = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n ws = new WebSocket(wsUrl);\n\n return new Promise((resolve, reject) => {\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 || data.stash || data.transcript || \"\", 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 || \"ASR error\");\n onError?.(err);\n reject(err);\n }\n };\n\n ws!.onerror = () => {\n const err = new Error(\"WebSocket error\");\n onError?.(err);\n reject(err);\n };\n\n ws!.onclose = () => {\n recorder?.stop();\n recorder = null;\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 recorder = await createRealtimeRecorder();\n await recorder.start((audio) => {\n if (!ws || ws.readyState !== WebSocket.OPEN) return;\n ws.send(\n JSON.stringify({\n type: \"input_audio_buffer.append\",\n audio,\n })\n );\n });\n }\n\n async function stopRecording() {\n try {\n await recorder?.stop();\n } catch (err) {}\n recorder = null;\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 ws?.close();\n ws = null;\n }\n\n return {\n connect,\n startRecording,\n stopRecording,\n close,\n };\n}\n"],"mappings":";AAIA,IAAM,WAAW;AAOjB,eAAe,yBAAoD;AACjE,MAAI;AACJ,MAAI;AACJ,MAAI;AACJ,MAAI;AAEJ,SAAO;AAAA,IACL,MAAM,MAAM,SAAS;AACnB,eAAS,MAAM,UAAU,aAAa,aAAa;AAAA,QACjD,OAAO,EAAE,YAAY,MAAO,cAAc,GAAG,kBAAkB,KAAK;AAAA,MACtE,CAAC;AAED,UAAI,+CAAY,SAAS;AACzB,UAAI,qGAAwB,SAAS;AAErC,YAAM,IAAI,aAAa,EAAE,YAAY,KAAM,CAAC;AAC5C,eAAS,IAAI,wBAAwB,MAAM;AAC3C,kBAAY,IAAI,sBAAsB,MAAM,GAAG,CAAC;AAEhD,gBAAU,iBAAiB,CAAC,MAAM;AAChC,cAAM,YAAY,EAAE,YAAY,eAAe,CAAC;AAChD,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,KAAK,CAAC,CAAC;AACrD,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,KAAK,CAAC;AAAA,QAC7C;AAEA,gBAAQ,KAAK,MAAM,CAAC;AAAA,MACtB;AAEA,aAAO,QAAQ,SAAS;AACxB,gBAAU,QAAQ,IAAI,WAAW;AAAA,IACnC;AAAA,IAEA,MAAM,OAAO;AACX,cAAQ,UAAU,EAAE,QAAQ,CAAC,MAAM,EAAE,KAAK,CAAC;AAC3C,cAAQ,WAAW;AACnB,iBAAW,WAAW;AACtB,YAAM,KAAK,MAAM;AAAA,IACnB;AAAA,EACF;AACF;AAEA,IAAM,MAAM,CAAC,SAAiB,OAAO,OAAO;AAC1C,UAAQ,IAAI,IAAI,IAAI,KAAK,OAAO;AAClC;AAoCO,SAAS,gBAAgB,QAAoC;AAClE,QAAM;AAAA,IACJ;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,IACA,OAAO;AAAA,IACP;AAAA,EACF,IAAI;AAEJ,MAAI,KAAuB;AAC3B,MAAI,WAAoC;AAExC,iBAAe,UAAyB;AAEtC,QAAI,QAAQ;AACZ,QAAI,gBAAgB;AAClB,YAAM,QAAQ,eAAe;AAC7B,UAAI,OAAO;AACT,cAAM,YAAY,KAAK,SAAS,GAAG,IAAI,MAAM;AAC7C,gBAAQ,GAAG,IAAI,GAAG,SAAS,SAAS,mBAAmB,KAAK,CAAC;AAAA,MAC/D;AAAA,IACF;AAEA,SAAK,IAAI,UAAU,KAAK;AAExB,WAAO,IAAI,QAAQ,CAAC,SAAS,WAAW;AACtC,SAAI,YAAY,CAAC,UAAU;AACzB,cAAM,OAAO,KAAK,MAAM,MAAM,IAAI;AAElC,YAAI,KAAK,SAAS,mBAAmB;AACnC,oBAAU;AACV,kBAAQ;AAAA,QACV;AAEA,YAAI,KAAK,SAAS,qCAAqC;AACrD,0BAAgB;AAAA,QAClB;AAEA,YAAI,KAAK,SAAS,qCAAqC;AACrD,wBAAc;AAAA,QAChB;AAEA,YAAI,KAAK,SAAS,oDAAoD;AACpE,yBAAe,KAAK,QAAQ,KAAK,SAAS,KAAK,cAAc,IAAI,KAAK;AAAA,QACxE;AAEA,YAAI,KAAK,SAAS,yDAAyD;AACzE,yBAAe,KAAK,QAAQ,KAAK,cAAc,IAAI,IAAI;AAAA,QACzD;AAEA,YAAI,KAAK,SAAS,SAAS;AACzB,gBAAM,MAAM,IAAI,MAAM,KAAK,OAAO,WAAW,WAAW;AACxD,oBAAU,GAAG;AACb,iBAAO,GAAG;AAAA,QACZ;AAAA,MACF;AAEA,SAAI,UAAU,MAAM;AAClB,cAAM,MAAM,IAAI,MAAM,iBAAiB;AACvC,kBAAU,GAAG;AACb,eAAO,GAAG;AAAA,MACZ;AAEA,SAAI,UAAU,MAAM;AAClB,kBAAU,KAAK;AACf,mBAAW;AACX,aAAK;AACL,kBAAU;AAAA,MACZ;AAAA,IACF,CAAC;AAAA,EACH;AAEA,iBAAe,iBAAgC;AAC7C,QAAI,CAAC,MAAM,GAAG,eAAe,UAAU,MAAM;AAC3C,YAAM,IAAI,MAAM,yBAAyB;AAAA,IAC3C;AAEA,eAAW,MAAM,uBAAuB;AACxC,UAAM,SAAS,MAAM,CAAC,UAAU;AAC9B,UAAI,CAAC,MAAM,GAAG,eAAe,UAAU,KAAM;AAC7C,SAAG;AAAA,QACD,KAAK,UAAU;AAAA,UACb,MAAM;AAAA,UACN;AAAA,QACF,CAAC;AAAA,MACH;AAAA,IACF,CAAC;AAAA,EACH;AAEA,iBAAe,gBAAgB;AAC7B,QAAI;AACF,YAAM,UAAU,KAAK;AAAA,IACvB,SAAS,KAAK;AAAA,IAAC;AACf,eAAW;AAEX,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,MAAM;AACV,SAAK;AAAA,EACP;AAEA,SAAO;AAAA,IACL;AAAA,IACA;AAAA,IACA;AAAA,IACA;AAAA,EACF;AACF;","names":[]}
1
+ {"version":3,"sources":["../src/asr-client.ts","../src/http-asr-client.ts"],"sourcesContent":["/**\n * ASR Realtime WebSocket Client\n *\n * WebSocket-based real-time speech recognition for streaming transcription.\n * Suitable for scenarios requiring real-time results like voice input, live captions, etc.\n *\n * @example\n * ```typescript\n * const client = createASRClient({\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 * // ... speak ...\n * client.stopRecording();\n * client.close();\n * ```\n */\n\nconst ASR_PATH = \"/api/proxy/builtin/platform/qwen-asr-realtime/api-ws/v1/realtime\";\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;\n let ctx: AudioContext;\n let source: MediaStreamAudioSourceNode;\n let processor: ScriptProcessorNode;\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 await ctx?.close();\n },\n };\n}\n\nconst log = (message: string, type = \"\") => {\n console.log(`[${type}]`, message);\n};\n\nexport interface ASRClientConfig {\n /** Audio format, default 'pcm16' */\n audioFormat?: \"pcm16\" | \"g711a\" | \"g711u\";\n /** Sample rate, default 16000 */\n sampleRate?: number;\n /** Get access token for WebSocket authentication */\n getAccessToken?: () => string | null;\n /** Called when connection is ready */\n onReady?: () => void;\n /** Called when speech is detected */\n onSpeechStart?: () => void;\n /** Called when speech stops */\n onSpeechEnd?: () => void;\n /** Called on transcript result */\n onTranscript?: (text: string, isFinal: boolean) => void;\n /** Called on error */\n onError?: (error: Error) => void;\n /** Called on close */\n onClose?: () => void;\n}\n\nexport interface ASRClient {\n /** Connect to ASR service */\n connect(): Promise<void>;\n /** Start recording from microphone */\n startRecording(): Promise<void>;\n /** Stop recording */\n stopRecording(): void;\n /** Close connection */\n close(): void;\n}\n\nfunction createASRClient(config: ASRClientConfig): ASRClient {\n const { onReady, onSpeechStart, onSpeechEnd, onTranscript, onError, onClose, getAccessToken } =\n config;\n\n let ws: WebSocket | null = null;\n let recorder: RealtimeRecorder | null = null;\n\n const path = ASR_PATH;\n\n async function connect(): Promise<void> {\n // Build WebSocket URL with optional token parameter\n let wsUrl = path;\n if (getAccessToken) {\n const token = getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n wsUrl = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n ws = new WebSocket(wsUrl);\n\n return new Promise((resolve, reject) => {\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 || data.stash || data.transcript || \"\", 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 || \"ASR error\");\n onError?.(err);\n reject(err);\n }\n };\n\n ws!.onerror = () => {\n const err = new Error(\"WebSocket error\");\n onError?.(err);\n reject(err);\n };\n\n ws!.onclose = () => {\n recorder?.stop();\n recorder = null;\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 recorder = await createRealtimeRecorder();\n await recorder.start((audio) => {\n if (!ws || ws.readyState !== WebSocket.OPEN) return;\n ws.send(\n JSON.stringify({\n type: \"input_audio_buffer.append\",\n audio,\n })\n );\n });\n }\n\n async function stopRecording() {\n try {\n await recorder?.stop();\n } catch (err) {}\n recorder = null;\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 ws?.close();\n ws = null;\n }\n\n return {\n connect,\n startRecording,\n stopRecording,\n close,\n };\n}\n\nexport default (authConfig: Pick<ASRClientConfig, \"getAccessToken\">) => (config: ASRClientConfig) =>\n createASRClient({ ...authConfig, ...config });\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 * // ... speak ...\n * const result = await client.stopRecording();\n * console.log(\"Result:\", result);\n * ```\n */\n\nconst ASR_HTTP_PATH = \"/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\ninterface Recorder {\n start(): Promise<void>;\n stop(): Promise<{ pcm: Int16Array; sampleRate: number }>;\n}\n\n// 创建录音器\nasync function createRecorder(): Promise<Recorder> {\n let stream: MediaStream;\n let ctx: AudioContext;\n let node: AudioWorkletNode;\n let source: MediaStreamAudioSourceNode;\n const chunks: Int16Array[] = [];\n\n return {\n async start() {\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], { type: \"application/javascript\" });\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\n async stop() {\n source?.disconnect();\n node?.disconnect();\n stream?.getTracks().forEach((t) => t.stop());\n if (ctx) {\n await ctx.close();\n }\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 return { pcm, sampleRate: ctx?.sampleRate ?? 16000 };\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 return new Blob([buffer], { type: \"audio/wav\" });\n}\n\nfunction blobToBase64(blob: Blob): Promise<string> {\n return new Promise((resolve, reject) => {\n const reader = new FileReader();\n reader.onloadend = () => {\n const result = reader.result as string;\n resolve(result.split(\",\")[1] || \"\");\n };\n reader.onerror = reject;\n reader.readAsDataURL(blob);\n });\n}\n\n/**\n * Recognize from Blob\n * @param blob - Audio Blob\n * @returns Recognized text\n */\nasync function recognizeBlob(blob: Blob, path?: string): Promise<string> {\n const base64 = await blobToBase64(blob);\n const response = await fetch(path ?? ASR_HTTP_PATH, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: 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,${base64}` },\n },\n ],\n },\n ],\n }),\n });\n\n const data = await response.json();\n return data.choices?.[0]?.message?.content || \"\";\n}\n\n/** Recognize audio file */\nexport async function recognizeFile(file: File | Blob, path?: string): Promise<string> {\n return recognizeBlob(file, path);\n}\n\nexport interface ASRHttpClientConfig {\n /** Get access token */\n getAccessToken?(): string | null;\n /** Language, default 'zh' */\n language?: string;\n /** Sample rate, default 16000 */\n sampleRate?: number;\n /** Called when recording starts */\n onRecordingStart?: () => void;\n /** Called when recording stops */\n onRecordingStop?: () => void;\n /** Called with recognition result */\n onResult?: (text: string) => void;\n /** Called on error */\n onError?: (error: Error) => void;\n}\n\nexport interface ASRHttpClient {\n /** Start recording (press-to-talk) */\n startRecording(): Promise<void>;\n /** Stop recording and get result */\n stopRecording(): Promise<string>;\n /** Record for specific duration then recognize */\n recordAndRecognize(durationMs: number): Promise<string>;\n /** Recognize audio file (File or Blob) */\n recognizeFile(file: File | Blob): Promise<string>;\n /** Recognize audio from URL */\n recognizeUrl(audioUrl: string): Promise<string>;\n}\n\nexport function createASRHttpClient(config: ASRHttpClientConfig): ASRHttpClient {\n let recorder: Recorder | null = null;\n\n let path = ASR_HTTP_PATH;\n\n if (config.getAccessToken) {\n const token = config.getAccessToken();\n if (token) {\n const separator = path.includes(\"?\") ? \"&\" : \"?\";\n path = `${path}${separator}token=${encodeURIComponent(token)}`;\n }\n }\n\n return {\n async startRecording() {\n recorder = await createRecorder();\n await recorder.start();\n config.onRecordingStart?.();\n },\n\n async stopRecording() {\n if (!recorder) throw new Error(\"Not recording\");\n const { pcm, sampleRate } = await recorder.stop();\n recorder = null;\n config.onRecordingStop?.();\n const text = await recognizeBlob(pcmToWav(pcm, sampleRate), path);\n config.onResult?.(text);\n return text;\n },\n\n async recordAndRecognize(ms: number) {\n await this.startRecording();\n await new Promise((r) => setTimeout(r, ms));\n return this.stopRecording();\n },\n\n recognizeFile: (file: File | Blob) => {\n return recognizeFile(file, path);\n },\n\n async recognizeUrl(url: string) {\n const res = await fetch(path, {\n method: \"POST\",\n headers: { \"Content-Type\": \"application/json\" },\n body: JSON.stringify({\n model: \"qwen3-asr-flash\",\n messages: [\n {\n role: \"user\",\n content: [{ type: \"input_audio\", input_audio: { url } }],\n },\n ],\n }),\n });\n const data = await res.json();\n return 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package/package.json CHANGED
@@ -1,6 +1,6 @@
1
1
  {
2
2
  "name": "@amaster.ai/asr-client",
3
- "version": "1.1.0-beta.49",
3
+ "version": "1.1.0-beta.50",
4
4
  "description": "Qwen ASR Realtime WebSocket client with microphone recording",
5
5
  "type": "module",
6
6
  "main": "./dist/index.cjs",