Refactor ASR module and remove deprecated AliAsrServer

- Introduced AsrRelatime class for real-time ASR using WebSocket.
- Removed AliAsrServer and related files from the aliyun provider.
- Updated base class for ASR to use WSServer for WebSocket connections.
- Added new test cases for the updated ASR functionality.
- Cleaned up unused imports and files across the project.
- Adjusted TypeScript configuration for better module resolution.
- Implemented silence generation for audio streaming.

src/asr/provider/aliyun/base.ts

@@ -1,42 +1,128 @@
-import RPCClient from '@alicloud/pop-core';
 
-interface TokenResponse {
-  Token: {
-    Id: string;
-    ExpireTime: number;
-  };
-}
-type AliCommonOptions = {
-  accessKeyId: string;
-  accessKeySecret: string;
-};
-export class AliCommon {
-  private accessKeyId: string;
-  private accessKeySecret: string;
-  private endpoint: string;
-  private apiVersion: string;
-  token = '';
-  expireTime = 0;
-  constructor(opts?: AliCommonOptions) {
-    this.accessKeyId = opts?.accessKeyId || process.env.ALIYUN_AK_ID || '';
-    this.accessKeySecret = opts?.accessKeySecret || process.env.ALIYUN_AK_SECRET || '';
-    this.endpoint = 'http://nls-meta.cn-shanghai.aliyuncs.com';
-    this.apiVersion = '2019-02-28';
+import { WSServer, WSSOptions } from '@kevisual/video-tools/asr/ws.ts';
+type Options = {
+  model?: string;
+  token?: string;
+} & Partial<WSSOptions>
+
+/**
+ * 阿里云实时语音识别服务
+ * new AsrRelatime({
+ *   token: 'your_token',
+ *   model: 'general_16k',
+ *   enableServerVad: true,
+ *   onConnect: async () => {
+ *     await asr.sendSessionUpdate();
+ *   }
+ * });
+ */
+export class AsrRelatime extends WSServer {
+  static baseURL = "wss://dashscope.aliyuncs.com/api-ws/v1/realtime"
+  /**
+   * 是否启用服务端VAD功能，true为VAD模式，false为Manual模式
+   */
+  enableServerVad: boolean = true;
+  constructor(options: Options) {
+    const { url: _, wsOptions: _wsOptions, ...rest } = options;
+    const wsOptions: WSSOptions['wsOptions'] = {
+      ..._wsOptions,
+      headers: {
+        Authorization: `Bearer ${options.token}`,
+        'OpenAi-Beta': 'realtime=v1',
+        ..._wsOptions?.headers
+      }
+    };
+    const models = options.model || 'qwen3-asr-flash-realtime';
+    const url = AsrRelatime.baseURL + `?model=${models}`;
+    super({ ...rest, url, wsOptions, onConnect: options.onConnect });
   }
-  async getToken() {
-    if (this.token && this.expireTime > Date.now()) {
-      return this.token;
+  async sendSessionUpdate() {
+    const { ws, enableServerVad } = this;
+    const connected = await this.checkConnected()
+    if (!connected) {
+      this.reconnect({ timeout: 60 * 1000 });
+      return;
+    }
+    const event = {
+      event_id: 'event_123',
+      type: 'session.update',
+      session: {
+        modalities: ['text'],
+        input_audio_format: 'pcm',
+        sample_rate: 16000,
+        input_audio_transcription: {
+          language: 'zh'
+        },
+        turn_detection: null
+      }
+    };
+    if (enableServerVad) {
+      event.session.turn_detection = {
+        type: 'server_vad',
+        threshold: 0.2,
+        silence_duration_ms: 800
+      }
+    }
+    ws.send(JSON.stringify(event));
+  }
+  async start() {
+    this.sendSessionUpdate();
+  }
+  async sendBuffer(buffer: Buffer) {
+    const { ws, enableServerVad } = this;;
+    const connected = await this.checkConnected()
+    if (!connected) {
+      this.reconnect({ timeout: 60 * 1000 });
+      return;
     }
-    const client = new RPCClient({
-      accessKeyId: this.accessKeyId,
-      accessKeySecret: this.accessKeySecret,
-      endpoint: this.endpoint,
-      apiVersion: this.apiVersion,
-    });
 
-    const result = await client.request<TokenResponse>('CreateToken', {});
-    this.token = result.Token.Id;
-    this.expireTime = result.Token.ExpireTime * 1000;
-    return result.Token.Id;
+    let offset = 0;
+    const bufferLength = Buffer.byteLength(buffer);
+    const chunkSize = 3200; // 约0.1s的PCM16音频 // max lenghth 262144
+    while (offset < bufferLength) {
+      const chunkBuffer = buffer.subarray(offset, offset + chunkSize);
+      offset += chunkSize;
+      const encoded = chunkBuffer.toString('base64');
+      const appendEvent = {
+        event_id: `event_${Date.now()}`,
+        type: 'input_audio_buffer.append',
+        audio: encoded
+      };
+      ws.send(JSON.stringify(appendEvent));
+    }
+    if (!enableServerVad) {
+      const commitEvent = {
+        event_id: 'event_789',
+        type: 'input_audio_buffer.commit'
+      };
+      ws.send(JSON.stringify(commitEvent));
+    }
+  }
+  async onMessage(event: MessageEvent) {
+    super.onMessage(event);
+    const data = event.data;
+    try {
+      const result = JSON.parse(data.toString());
+      const isEnd = await this.isEnd(result.type);
+      if (isEnd && result?.transcript) {
+        const text = result.transcript;
+        this.emitter.emit('result', {
+          text: text,
+          raw: result
+        });
+      }
+    } catch (error) {
+      console.log('error', error);
+    }
+  }
+  async isEnd(type: string) {
+    const types = ['conversation.item.input_audio_transcription.text', 'conversation.item.input_audio_transcription.completed'];
+    if (type === types[1]) {
+      return true;
+    }
+    return false;
+  }
+  async sendBlank(buffer?: Buffer): Promise<void> {
+    this.sendBuffer(buffer || this.generateSilence(2));
   }
 }