STT流式输入业务逻辑处理
This commit is contained in:
6
pom.xml
6
pom.xml
@@ -304,6 +304,12 @@
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<version>4.5.14</version>
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</dependency>
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<dependency>
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<groupId>org.java-websocket</groupId>
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<artifactId>Java-WebSocket</artifactId>
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<version>1.5.4</version> <!-- 最新版本可到 Maven 仓库查询 -->
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</dependency>
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</dependencies>
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</dependencyManagement>
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@@ -1,6 +1,7 @@
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package com.vetti.socket;
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import cn.hutool.core.util.StrUtil;
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import cn.hutool.json.JSONObject;
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import cn.hutool.json.JSONUtil;
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import com.vetti.common.ai.elevenLabs.ElevenLabsClient;
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import com.vetti.common.ai.gpt.OpenAiStreamClient;
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@@ -9,9 +10,15 @@ import com.vetti.common.ai.whisper.WhisperClient;
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import com.vetti.common.config.RuoYiConfig;
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import com.vetti.common.utils.spring.SpringUtils;
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import lombok.extern.slf4j.Slf4j;
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import okhttp3.*;
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import org.apache.commons.io.FileUtils;
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import org.springframework.beans.factory.annotation.Value;
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import org.springframework.stereotype.Component;
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import javax.sound.sampled.AudioFormat;
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import javax.sound.sampled.AudioSystem;
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import javax.sound.sampled.DataLine;
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import javax.sound.sampled.TargetDataLine;
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import javax.websocket.*;
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import javax.websocket.server.PathParam;
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import javax.websocket.server.ServerEndpoint;
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@@ -19,9 +26,11 @@ import java.io.File;
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import java.io.FileOutputStream;
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import java.io.IOException;
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import java.nio.ByteBuffer;
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import java.util.Base64;
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import java.util.HashMap;
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import java.util.Map;
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import java.util.concurrent.ConcurrentHashMap;
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import java.util.concurrent.CountDownLatch;
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/**
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* 语音面试 web处理器
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@@ -31,11 +40,41 @@ import java.util.concurrent.ConcurrentHashMap;
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@Component
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public class ChatWebSocketHandler {
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@Value("${whisper.apiUrl}")
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private String API_URL;
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@Value("${whisper.model}")
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private String MODEL;
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@Value("${whisper.apiKey}")
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private String apiKey;
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@Value("${whisper.language}")
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private String language;
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/**
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* 16kHz
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*/
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private static final int SAMPLE_RATE = 16000;
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/**
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* 4 KB 每次读取
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*/
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private static final int BUFFER_SIZE = 4096;
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/**
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* 每样本 16 位
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*/
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private static final int BITS_PER_SAMPLE = 16;
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/**
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* 缓存客户端流式解析的语音文本数据
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*/
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private final Map<String,String> cacheClientTts = new ConcurrentHashMap<>();
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/**
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* 缓存客户端调用OpenAi中的websocket-STT 流式传输数据
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*/
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private final Map<String, WebSocket> cacheWebSocket = new ConcurrentHashMap<>();
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// 语音文件保存目录
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private static final String VOICE_STORAGE_DIR = "/voice_files/";
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@@ -61,6 +100,8 @@ public class ChatWebSocketHandler {
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public void onOpen(Session session, @PathParam("clientId") String clientId) {
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log.info("WebSocket 链接已建立:{}", clientId);
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cacheClientTts.put(clientId,new String());
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//初始化STT流式语音转换文本的socket链接
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createWhisperRealtimeSocket(clientId);
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}
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// 接收文本消息
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@@ -148,20 +189,18 @@ public class ChatWebSocketHandler {
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log.info("3、开始接收数据流时间:{}",System.currentTimeMillis()/1000);
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try{
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//接收到数据流后直接就进行SST处理
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//拿到文件进行文字转换
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saveAsWebM(bytes,pathUrl);
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WhisperClient whisperClient = SpringUtils.getBean(WhisperClient.class);
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String resultText = whisperClient.handleVoiceToText(pathUrl);
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log.info("STT:{}",resultText);
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//进行客户端文本数据存储
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String cacheString = cacheClientTts.get(clientId);
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if(StrUtil.isNotEmpty(cacheString)){
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cacheString = cacheString+resultText;
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}else {
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cacheString = resultText;
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//发送消息
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WebSocket webSocket = cacheWebSocket.get(clientId);
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if(webSocket != null){
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log.info("3.1 开始发送数据音频流啦");
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// 将音频数据转换为 Base64 编码的字符串
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String base64Audio = Base64.getEncoder().encodeToString(bytes);
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String message = "{ \"type\": \"input_audio_buffer.append\", \"audio\": \"" + base64Audio + "\" }";
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webSocket.send(message);
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// 3. 提交音频并请求转录
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webSocket.send("{\"type\": \"input_audio_buffer.commit\"}");
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webSocket.send("{\"type\": \"response.create\"}");
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}
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cacheClientTts.put(clientId,cacheString);
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}catch (Exception e){
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e.printStackTrace();
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}
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@@ -243,5 +282,87 @@ public class ChatWebSocketHandler {
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return null;
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}
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/**
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* 创建STT WebSocket 客户端链接
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* @param clientId 客户端ID
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*/
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private void createWhisperRealtimeSocket(String clientId){
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try{
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OkHttpClient client = new OkHttpClient();
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CountDownLatch latch = new CountDownLatch(1);
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// 设置 WebSocket 请求
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Request request = new Request.Builder()
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.url(API_URL)
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.addHeader("Authorization", "Bearer " + apiKey)
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.addHeader("OpenAI-Beta", "realtime=v1")
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.build();
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client.newWebSocket(request, new WebSocketListener() {
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@Override
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public void onOpen(WebSocket webSocket, Response response) {
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System.out.println("✅ WebSocket 连接成功");
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//发送配置
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JSONObject config = new JSONObject();
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JSONObject sessionConfig = new JSONObject();
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JSONObject transcription = new JSONObject();
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JSONObject turnDetection = new JSONObject();
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// 配置转录参数
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transcription.put("model", "gpt-4o-mini-transcribe");
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transcription.put("language", language); // 中文
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// 配置断句检测
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turnDetection.put("type", "server_vad");
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turnDetection.put("prefix_padding_ms", 300);
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turnDetection.put("silence_duration_ms", 10);
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// 组装完整配置
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sessionConfig.put("input_audio_transcription", transcription);
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sessionConfig.put("turn_detection", turnDetection);
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config.put("type", "transcription_session.update");
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config.put("session", sessionConfig);
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webSocket.send(config.toString());
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// 1. 启动音频缓冲
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webSocket.send("{\"type\": \"input_audio_buffer.start\"}");
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//存储客户端webSocket对象,对数据进行隔离处理
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cacheWebSocket.put(clientId,webSocket);
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}
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@Override
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public void onMessage(WebSocket webSocket, String text) {
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System.out.println("📩 收到转录结果: " + text);
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//对数据进行解析
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if(StrUtil.isNotEmpty(text)){
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Map<String,String> mapResultData = JSONUtil.toBean(text,Map.class);
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if("conversation.item.input_audio_transcription.delta".equals(mapResultData.get("type"))){
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String resultText = mapResultData.get("delta");
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//进行客户端文本数据存储
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String cacheString = cacheClientTts.get(clientId);
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if(StrUtil.isNotEmpty(cacheString)){
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cacheString = cacheString+resultText;
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}else {
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cacheString = resultText;
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}
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cacheClientTts.put(clientId,cacheString);
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}
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}
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}
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@Override
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public void onFailure(WebSocket webSocket, Throwable t, Response response) {
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System.err.println("❌ 连接失败: " + t.getMessage());
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latch.countDown();
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}
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@Override
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public void onClosing(WebSocket webSocket, int code, String reason) {
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System.out.println("⚠️ 连接即将关闭: " + reason);
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webSocket.close(1000, null);
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latch.countDown();
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}
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});
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// 等待 WebSocket 关闭
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latch.await();
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}catch (Exception e){
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e.printStackTrace();
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}
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}
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}
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@@ -165,6 +165,15 @@
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<artifactId>httpmime</artifactId>
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</dependency>
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<dependency>
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<groupId>org.java-websocket</groupId>
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<artifactId>Java-WebSocket</artifactId>
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</dependency>
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<dependency>
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<groupId>org.apache.tomcat.embed</groupId>
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<artifactId>tomcat-embed-websocket</artifactId>
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</dependency>
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</dependencies>
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@@ -0,0 +1,165 @@
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package com.vetti.common.ai.whisper;
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import cn.hutool.json.JSONObject;
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import lombok.extern.slf4j.Slf4j;
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import org.java_websocket.client.WebSocketClient;
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import org.java_websocket.handshake.ServerHandshake;
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import java.net.URI;
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import java.net.URISyntaxException;
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import java.nio.ByteBuffer;
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import java.util.Base64;
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import java.util.HashMap;
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import java.util.Map;
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@Slf4j
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public class OpenAIRealtimeClient extends WebSocketClient {
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// 构造方法:初始化连接地址和请求头(携带认证信息)
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public OpenAIRealtimeClient(String apiKey) throws URISyntaxException {
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super(
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new URI("wss://api.openai.com/v1/realtime?intent=transcription"),
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buildHeaders(apiKey) // 构建请求头
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);
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}
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// 构建请求头(携带认证和内容类型)
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private static Map<String, String> buildHeaders(String apiKey) {
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Map<String, String> headers = new HashMap<>();
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headers.put("Authorization", "Bearer " + apiKey); // 核心认证头
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headers.put("Content-Type", "application/json"); // 根据接口要求调整
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headers.put("OpenAI-Beta", "realtime=v1"); // 若接口要求 beta 版本标识
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return headers;
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}
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// 连接成功回调
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@Override
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public void onOpen(ServerHandshake handshakedata) {
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System.out.println("WebSocket 连接已打开,状态码:" + handshakedata.getHttpStatus());
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// 连接成功后可发送初始化消息(如配置转录参数)
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sendInitMessage();
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}
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// 接收服务器消息回调(处理转录结果)
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@Override
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public void onMessage(String message) {
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System.out.println("收到转录文本:" + message);
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// 解析 JSON 格式的转录结果(可使用 Jackson/Gson 等库)
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}
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// 接收二进制消息(若服务器返回二进制数据,如音频片段确认)
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@Override
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public void onMessage(ByteBuffer bytes) {
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System.out.println("收到二进制数据,长度:" + bytes.remaining());
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// 处理二进制消息(如需)
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}
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// 连接关闭回调
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@Override
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public void onClose(int code, String reason, boolean remote) {
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System.out.println("连接关闭,状态码:" + code + ",原因:" + reason);
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}
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// 连接错误回调
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@Override
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public void onError(Exception ex) {
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System.err.println("连接错误:" + ex.getMessage());
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ex.printStackTrace();
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}
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// 发送初始化消息(根据 OpenAI 接口要求配置转录参数)
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private void sendInitMessage() {
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JSONObject config = new JSONObject();
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JSONObject sessionConfig = new JSONObject();
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JSONObject transcription = new JSONObject();
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JSONObject turnDetection = new JSONObject();
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// 配置转录参数
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transcription.put("model", "gpt-4o-mini-transcribe");
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transcription.put("language", "zh"); // 中文
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// 配置断句检测
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turnDetection.put("type", "server_vad");
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turnDetection.put("prefix_padding_ms", 300);
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turnDetection.put("silence_duration_ms", 10);
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// 组装完整配置
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sessionConfig.put("input_audio_transcription", transcription);
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sessionConfig.put("turn_detection", turnDetection);
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config.put("type", "transcription_session.update");
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config.put("session", sessionConfig);
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this.send(config.toString());
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System.out.println("已发送初始化配置");
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}
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// 发送音频数据(核心:将麦克风/文件的音频流发送到服务器)
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public void sendAudioData(byte[] audioBytes) {
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if (this.isOpen()) {
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// 按接口要求封装音频数据(通常为 JSON 包裹二进制,或直接发送二进制)
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// OpenAI要求语音数据以Base64编码发送
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String base64Chunk = Base64.getEncoder().encodeToString(audioBytes);
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String audioJson = "{\n" +
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" \"type\": \"input_audio_buffer.append\",\n" +
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" \"audio\": \""+base64Chunk+"\"\n" +
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"}";
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this.send(audioJson);
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// this.send(audioBytes);
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System.out.println("已发送音频数据,长度:" + audioBytes.length);
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} else {
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System.err.println("连接未打开,无法发送音频");
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}
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}
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public void commitData() {
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String base64Chunk = Base64.getEncoder().encodeToString(new byte[0]);
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String audioJson = "{\n" +
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" \"type\": \"input_audio_buffer.append\",\n" +
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" \"audio\": \""+base64Chunk+"\"\n" +
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"}";
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this.send(audioJson);
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}
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public static void main(String[] args) {
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String apiKey = "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"; // 替换为你的 OpenAI API Key
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try {
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// 创建客户端
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OpenAIRealtimeClient client = new OpenAIRealtimeClient(apiKey);
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// 连接服务器
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client.connectBlocking(); // 阻塞式连接(也可使用非阻塞 connect())
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// 模拟发送音频数据(实际应从麦克风或文件读取)
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// 注意:音频格式需符合接口要求(通常为 PCM 16kHz 单声道等)
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// 读取本地PCM文件(16kHz单声道16位)并分片发送
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try (java.io.FileInputStream fis = new java.io.FileInputStream("/Users/wangxiangshun/Desktop/临时文件/output1112.mp3")) {
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byte[] buffer = new byte[6400]; // 200ms的PCM数据(16000Hz*16位*1声道=32000字节/秒 → 6400字节/200ms)
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int len;
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while ((len = fis.read(buffer)) != -1) {
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byte[] chunk = new byte[len];
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System.arraycopy(buffer, 0, chunk, 0, len);
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client.sendAudioData(chunk);
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Thread.sleep(200); // 模拟实时流(每200ms发送一次)
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}
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}
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//发送一个空的二进制流
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// 4. 发送结束标记(空二进制消息)
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client.sendAudioData(new byte[600]);
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// 等待转录完成(实际场景需根据业务逻辑控制)
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Thread.sleep(20000);
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// 关闭连接
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client.close();
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} catch (Exception e) {
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e.printStackTrace();
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}
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}
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// 模拟读取音频数据(实际需用音频库采集,如 Java Sound API 或 VLCJ)
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private static byte[] readAudioFromSource() {
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// 示例:返回空字节数组(实际应填充真实音频数据)
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return new byte[1024];
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}
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}
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@@ -0,0 +1,115 @@
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package com.vetti.common.ai.whisper;
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import cn.hutool.json.JSONObject;
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import okhttp3.*;
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import javax.sound.sampled.AudioFormat;
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import javax.sound.sampled.AudioSystem;
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import javax.sound.sampled.DataLine;
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import javax.sound.sampled.TargetDataLine;
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import java.util.Base64;
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import java.util.concurrent.CountDownLatch;
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public class RealtimeTranscriptionMicrophone {
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private static final String API_KEY = "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";
|
||||
private static final String URL = "wss://api.openai.com/v1/realtime?intent=transcription";
|
||||
private static final int SAMPLE_RATE = 16000; // 16kHz
|
||||
private static final int BUFFER_SIZE = 4096; // 4 KB 每次读取
|
||||
private static final int BITS_PER_SAMPLE = 16; // 每样本 16 位
|
||||
|
||||
public static void main(String[] args) throws Exception {
|
||||
|
||||
OkHttpClient client = new OkHttpClient();
|
||||
CountDownLatch latch = new CountDownLatch(1);
|
||||
|
||||
// 设置 WebSocket 请求
|
||||
Request request = new Request.Builder()
|
||||
.url(URL)
|
||||
.addHeader("Authorization", "Bearer " + API_KEY)
|
||||
.addHeader("OpenAI-Beta", "realtime=v1")
|
||||
.build();
|
||||
|
||||
WebSocket ws = client.newWebSocket(request, new WebSocketListener() {
|
||||
@Override
|
||||
public void onOpen(WebSocket webSocket, Response response) {
|
||||
System.out.println("✅ WebSocket 连接成功");
|
||||
|
||||
//发送配置
|
||||
JSONObject config = new JSONObject();
|
||||
JSONObject sessionConfig = new JSONObject();
|
||||
JSONObject transcription = new JSONObject();
|
||||
JSONObject turnDetection = new JSONObject();
|
||||
|
||||
// 配置转录参数
|
||||
transcription.put("model", "gpt-4o-mini-transcribe");
|
||||
transcription.put("language", "zh"); // 中文
|
||||
// 配置断句检测
|
||||
turnDetection.put("type", "server_vad");
|
||||
turnDetection.put("prefix_padding_ms", 300);
|
||||
turnDetection.put("silence_duration_ms", 10);
|
||||
// 组装完整配置
|
||||
sessionConfig.put("input_audio_transcription", transcription);
|
||||
sessionConfig.put("turn_detection", turnDetection);
|
||||
config.put("type", "transcription_session.update");
|
||||
config.put("session", sessionConfig);
|
||||
|
||||
webSocket.send(config.toString());
|
||||
|
||||
// 1. 启动音频缓冲
|
||||
webSocket.send("{\"type\": \"input_audio_buffer.start\"}");
|
||||
|
||||
// 2. 开始录音并实时发送
|
||||
new Thread(() -> {
|
||||
try {
|
||||
// 设置麦克风输入流
|
||||
AudioFormat format = new AudioFormat(SAMPLE_RATE, BITS_PER_SAMPLE, 1, true, false);
|
||||
DataLine.Info info = new DataLine.Info(TargetDataLine.class, format);
|
||||
TargetDataLine line = (TargetDataLine) AudioSystem.getLine(info);
|
||||
line.open(format);
|
||||
line.start();
|
||||
|
||||
byte[] buffer = new byte[BUFFER_SIZE];
|
||||
int bytesRead;
|
||||
while ((bytesRead = line.read(buffer, 0, buffer.length)) > 0) {
|
||||
// 将音频数据转换为 Base64 编码的字符串
|
||||
byte[] audioData = new byte[bytesRead];
|
||||
System.arraycopy(buffer, 0, audioData, 0, bytesRead);
|
||||
String base64Audio = Base64.getEncoder().encodeToString(audioData);
|
||||
String message = "{ \"type\": \"input_audio_buffer.append\", \"audio\": \"" + base64Audio + "\" }";
|
||||
webSocket.send(message);
|
||||
}
|
||||
|
||||
// 3. 提交音频并请求转录
|
||||
webSocket.send("{\"type\": \"input_audio_buffer.commit\"}");
|
||||
webSocket.send("{\"type\": \"response.create\"}");
|
||||
} catch (Exception e) {
|
||||
e.printStackTrace();
|
||||
}
|
||||
}).start();
|
||||
}
|
||||
|
||||
@Override
|
||||
public void onMessage(WebSocket webSocket, String text) {
|
||||
System.out.println("📩 收到转录结果: " + text);
|
||||
}
|
||||
|
||||
@Override
|
||||
public void onFailure(WebSocket webSocket, Throwable t, Response response) {
|
||||
System.err.println("❌ 连接失败: " + t.getMessage());
|
||||
latch.countDown();
|
||||
}
|
||||
|
||||
@Override
|
||||
public void onClosing(WebSocket webSocket, int code, String reason) {
|
||||
System.out.println("⚠️ 连接即将关闭: " + reason);
|
||||
webSocket.close(1000, null);
|
||||
latch.countDown();
|
||||
}
|
||||
});
|
||||
|
||||
// 等待 WebSocket 关闭
|
||||
latch.await();
|
||||
}
|
||||
}
|
||||
|
||||
Reference in New Issue
Block a user