Vue实现AI问答小助手(3)：录音与语音转文字全流程指南

2025年11月14日 互联网

一、技术选型与前期准备

1.1 浏览器原生API分析

现代浏览器提供了MediaRecorder和Web Speech API两大核心接口：

  • MediaRecorder API：支持实时音频流捕获，兼容Chrome/Firefox/Edge等主流浏览器
  • Web Speech API：包含语音识别(SpeechRecognition)和语音合成(SpeechSynthesis)模块

但需注意：

  • Safari对部分API的支持存在缺陷（如MediaRecorder的opus编码）
  • 移动端浏览器权限管理更为严格

1.2 第三方服务对比

服务类型 优势 限制条件
浏览器原生API 零依赖，数据不出域 功能有限，移动端兼容性差
WebSocket SDK 支持高并发，低延迟 需要后端服务支持
商业ASR服务 准确率高，支持多语言 调用次数限制，可能产生费用

推荐组合方案：

  • 基础功能使用Web Speech API
  • 高精度需求接入阿里云/腾讯云ASR服务

二、核心功能实现

2.1 录音组件开发

2.1.1 权限申请与设备检测

  1. async function checkAudioPermission() {
  2.   try {
  3.     const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
  4.     stream.getTracks().forEach(track => track.stop());
  5.     return { status: 'granted', message: '麦克风访问权限已获取' };
  6.   } catch (err) {
  7.     return { 
  8.       status: 'denied', 
  9.       message: `权限错误: ${err.message}`,
  10.       code: err.name === 'NotAllowedError' ? 403 : 500
  11.     };
  12.   }
  13. }

2.1.2 录音状态管理

采用Vue3的Composition API实现响应式控制：

  1. import { ref, onUnmounted } from 'vue';
  3. export function useAudioRecorder() {
  4.   const isRecording = ref(false);
  5.   const mediaRecorder = ref(null);
  6.   const audioChunks = ref([]);
  8.   const startRecording = async () => {
  9.     try {
  10.       const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
  11.       mediaRecorder.value = new MediaRecorder(stream, {
  12.         mimeType: 'audio/webm',
  13.         audioBitsPerSecond: 128000
  14.       });
  16.       mediaRecorder.value.ondataavailable = (event) => {
  17.         audioChunks.value.push(event.data);
  18.       };
  20.       mediaRecorder.value.start(100); // 每100ms收集一次数据
  21.       isRecording.value = true;
  22.     } catch (error) {
  23.       console.error('录音启动失败:', error);
  24.     }
  25.   };
  27.   const stopRecording = () => {
  28.     return new Promise((resolve) => {
  29.       if (!mediaRecorder.value) return resolve(null);
  31.       mediaRecorder.value.onstop = () => {
  32.         const audioBlob = new Blob(audioChunks.value, { type: 'audio/webm' });
  33.         resolve(audioBlob);
  34.         audioChunks.value = [];
  35.       };
  37.       mediaRecorder.value.stop();
  38.       isRecording.value = false;
  39.     });
  40.   };
  42.   onUnmounted(() => {
  43.     if (mediaRecorder.value?.state === 'recording') {
  44.       mediaRecorder.value.stop();
  45.     }
  46.   });
  48.   return { isRecording, startRecording, stopRecording };
  49. }

2.2 语音转文字实现

2.2.1 浏览器原生方案

  1. export function useSpeechRecognition() {
  2.   const recognition = ref(null);
  3.   const isListening = ref(false);
  4.   const transcript = ref('');
  6.   const initRecognition = () => {
  7.     recognition.value = new (window.SpeechRecognition || 
  8.                          window.webkitSpeechRecognition)();
  10.     recognition.value.continuous = true;
  11.     recognition.value.interimResults = true;
  12.     recognition.value.lang = 'zh-CN';
  14.     recognition.value.onresult = (event) => {
  15.       let interimTranscript = '';
  16.       for (let i = event.resultIndex; i < event.results.length; i++) {
  17.         const transcriptPiece = event.results[i][0].transcript;
  18.         if (event.results[i].isFinal) {
  19.           transcript.value += transcriptPiece;
  20.         } else {
  21.           interimTranscript += transcriptPiece;
  22.         }
  23.       }
  24.       // 这里可以添加实时显示中间结果的逻辑
  25.     };
  27.     recognition.value.onerror = (event) => {
  28.       console.error('识别错误:', event.error);
  29.     };
  30.   };
  32.   const startListening = () => {
  33.     if (!recognition.value) initRecognition();
  34.     recognition.value.start();
  35.     isListening.value = true;
  36.   };
  38.   const stopListening = () => {
  39.     if (recognition.value) {
  40.       recognition.value.stop();
  41.       isListening.value = false;
  42.     }
  43.   };
  45.   return { isListening, transcript, startListening, stopListening };
  46. }

2.2.2 云端ASR集成（以WebSocket为例）

  1. async function connectToASRService(audioBlob) {
  2.   const socket = new WebSocket('wss://asr.example.com/api');
  3.   const audioUrl = URL.createObjectURL(audioBlob);
  5.   return new Promise((resolve, reject) => {
  6.     socket.onopen = () => {
  7.       // 发送音频元数据
  8.       socket.send(JSON.stringify({
  9.         type: 'metadata',
  10.         format: 'webm',
  11.         sampleRate: 16000
  12.       }));
  14.       // 分段发送音频数据
  15.       const audioContext = new AudioContext();
  16.       fetch(audioUrl)
  17.         .then(response => response.arrayBuffer())
  18.         .then(buffer => audioContext.decodeAudioData(buffer))
  19.         .then(audioBuffer => {
  20.           const channelData = audioBuffer.getChannelData(0);
  21.           const sampleRate = audioBuffer.sampleRate;
  22.           const chunkSize = sampleRate * 0.5; // 每0.5秒发送一次
  24.           for (let i = 0; i < channelData.length; i += chunkSize) {
  25.             const chunk = channelData.slice(i, i + chunkSize);
  26.             const float32Array = new Float32Array(chunk);
  27.             socket.send(float32Array);
  28.           }
  30.           socket.send(JSON.stringify({ type: 'end' }));
  31.         });
  32.     };
  34.     socket.onmessage = (event) => {
  35.       const data = JSON.parse(event.data);
  36.       if (data.type === 'partial') {
  37.         // 实时更新部分识别结果
  38.       } else if (data.type === 'final') {
  39.         resolve(data.text);
  40.       }
  41.     };
  43.     socket.onerror = (error) => {
  44.       reject(new Error(`ASR连接错误: ${error}`));
  45.     };
  46.   });
  47. }

三、性能优化与错误处理

3.1 音频处理优化

  • 采样率转换：使用offlineAudioContext进行实时降采样

    1. function resampleAudio(audioBuffer, targetRate) {
    2. const offlineCtx = new OfflineAudioContext(
    3.   1, 
    4.   audioBuffer.length * targetRate / audioBuffer.sampleRate, 
    5.   targetRate
    6. );
    7. const source = offlineCtx.createBufferSource();
    8. source.buffer = audioBuffer;
    9. source.connect(offlineCtx.destination);
    10. source.start();
    11. return offlineCtx.startRendering();
    12. }

  • WebAssembly加速：使用librosa.js等WASM库进行特征提取

3.2 错误恢复机制

  1. const retryPolicy = {
  2.   maxRetries: 3,
  3.   timeout: 5000,
  4.   shouldRetry: (error) => {
  5.     return error.code !== 'NetworkError' || 
  6.            (error.message.includes('timeout') && retryCount < 2);
  7.   }
  8. };
  10. async function withRetry(fn, policy) {
  11.   let lastError = null;
  12.   for (let i = 0; i < policy.maxRetries; i++) {
  13.     try {
  14.       return await Promise.race([
  15.         fn(),
  16.         new Promise((_, reject) => 
  17.           setTimeout(() => reject(new Error('请求超时')), policy.timeout)
  18.         )
  19.       ]);
  20.     } catch (error) {
  21.       lastError = error;
  22.       if (!policy.shouldRetry(error)) break;
  23.       await new Promise(resolve => setTimeout(resolve, 1000 * (i + 1)));
  24.     }
  25.   }
  26.   throw lastError;
  27. }

四、完整组件示例

  1. <template>
  2.   <div class="voice-assistant">
  3.     <div class="control-panel">
  4.       <button @click="toggleRecording" :disabled="isProcessing">
  5.         {{ isRecording ? '停止录音' : '开始录音' }}
  6.       </button>
  7.       <button @click="toggleListening" :disabled="isProcessing">
  8.         {{ isListening ? '停止识别' : '语音识别' }}
  9.       </button>
  10.     </div>
  12.     <div class="status-display">
  13.       <div v-if="error" class="error-message">{{ error }}</div>
  14.       <div v-else-if="isProcessing" class="processing-indicator">
  15.         处理中... {{ progress }}%
  16.       </div>
  17.       <div v-else-if="transcript" class="transcript-display">
  18.         识别结果: {{ transcript }}
  19.       </div>
  20.     </div>
  21.   </div>
  22. </template>
  24. <script setup>
  25. import { ref } from 'vue';
  26. import { useAudioRecorder } from './composables/audioRecorder';
  27. import { useSpeechRecognition } from './composables/speechRecognition';
  29. const { isRecording, startRecording, stopRecording } = useAudioRecorder();
  30. const { isListening, transcript, startListening, stopListening } = 
  31.   useSpeechRecognition();
  33. const isProcessing = ref(false);
  34. const error = ref(null);
  35. const progress = ref(0);
  37. const toggleRecording = async () => {
  38.   if (isRecording.value) {
  39.     isProcessing.value = true;
  40.     progress.value = 0;
  42.     try {
  43.       const audioBlob = await stopRecording();
  44.       // 这里可以添加进度更新逻辑
  45.       const result = await processAudio(audioBlob);
  46.       transcript.value = result;
  47.     } catch (err) {
  48.       error.value = `处理失败: ${err.message}`;
  49.     } finally {
  50.       isProcessing.value = false;
  51.     }
  52.   } else {
  53.     await startRecording();
  54.   }
  55. };
  57. const toggleListening = () => {
  58.   if (isListening.value) {
  59.     stopListening();
  60.   } else {
  61.     startListening();
  62.   }
  63. };
  65. async function processAudio(audioBlob) {
  66.   // 实际项目中这里调用ASR服务
  67.   return new Promise(resolve => {
  68.     setTimeout(() => resolve('这是模拟的识别结果'), 1500);
  69.   });
  70. }
  71. </script>
  73. <style scoped>
  74. .voice-assistant {
  75.   max-width: 600px;
  76.   margin: 0 auto;
  77.   padding: 20px;
  78. }
  79. .control-panel {
  80.   display: flex;
  81.   gap: 10px;
  82.   margin-bottom: 20px;
  83. }
  84. .status-display {
  85.   min-height: 100px;
  86.   padding: 15px;
  87.   border: 1px solid #eee;
  88.   border-radius: 5px;
  89. }
  90. </style>

五、部署注意事项

  1. HTTPS要求：所有媒体API必须在安全上下文中使用
  2. 移动端适配：
    • iOS需要用户交互后才能访问麦克风
    • Android Chrome 70+支持完整功能
  3. 性能监控：
    • 使用PerformanceObserver监控音频处理耗时
    • 记录ASR服务的响应时间和准确率

本文提供的方案经过实际项目验证，在Chrome 115+和Firefox 114+环境下测试通过。开发者可根据实际需求选择原生API或混合方案，对于企业级应用建议采用专业ASR服务以获得更好的识别效果和稳定性。

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