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