456 lines
19 KiB
Java
456 lines
19 KiB
Java
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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import com.vetti.common.ai.gpt.service.OpenAiStreamListenerService;
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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.*;
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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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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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import javax.sound.sampled.*;
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import java.io.*;
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import java.nio.ByteBuffer;
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/**
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* 语音面试 web处理器
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*/
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@Slf4j
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@ServerEndpoint("/voice-websocket/{clientId}")
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@Component
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public class ChatWebSocketHandler {
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// @Value("${whisper.apiUrl}")
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private String API_URL = "wss://api.openai.com/v1/realtime?intent=transcription";
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// @Value("${whisper.model}")
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private String MODEL = "gpt-4o-mini-transcribe";
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// @Value("${whisper.apiKey}")
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private String apiKey = "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";
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// @Value("${whisper.language}")
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private String language = "en";
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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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// 语音结果文件保存目录
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private static final String VOICE_STORAGE_RESULT_DIR = "/voice_result_files/";
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public ChatWebSocketHandler() {
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// 初始化存储目录
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File dir = new File(RuoYiConfig.getProfile() + VOICE_STORAGE_DIR);
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if (!dir.exists()) {
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dir.mkdirs();
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}
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File resultDir = new File(RuoYiConfig.getProfile() + VOICE_STORAGE_RESULT_DIR);
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if (!resultDir.exists()) {
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resultDir.mkdirs();
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}
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}
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// 连接建立时调用
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@OnOpen
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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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@OnMessage
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public void onTextMessage(Session session, String message,@PathParam("clientId") String clientId) {
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System.out.println("接收到文本消息: " + message);
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try {
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//处理文本结果
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if(StrUtil.isNotEmpty(message)){
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Map<String,String> mapResult = JSONUtil.toBean(JSONUtil.parseObj(message),Map.class);
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String resultFlag = mapResult.get("msg");
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if("done".equals(resultFlag)){
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//发送消息
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WebSocket webSocket = cacheWebSocket.get(clientId);
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webSocket.send("{\"type\": \"input_audio_buffer.commit\"}");
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webSocket.send("{\"type\": \"response.create\"}");
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// if(webSocket != null){
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// webSocket.close(1000,null);
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// }
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//语音结束,开始进行回答解析
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String cacheResultText = cacheClientTts.get(clientId);
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log.info("1、开始进行AI回答时间:{}",System.currentTimeMillis()/1000);
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//把提问的文字发送给CPT(流式处理)
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OpenAiStreamClient aiStreamClient = SpringUtils.getBean(OpenAiStreamClient.class);
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aiStreamClient.streamChat(cacheResultText, new OpenAiStreamListenerService() {
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@Override
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public void onMessage(String content) {
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log.info("返回AI结果：{}",content.trim());
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// 实时输出内容
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//开始进行语音输出-流式持续输出
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log.info("2、开始进行AI回答时间:{}",System.currentTimeMillis()/1000);
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//把结果文字转成语音文件
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//生成文件
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//生成唯一文件名
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String resultFileName = clientId + "_" + System.currentTimeMillis() + ".wav";
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String resultPathUrl = RuoYiConfig.getProfile() + VOICE_STORAGE_RESULT_DIR + resultFileName;
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ElevenLabsClient elevenLabsClient = SpringUtils.getBean(ElevenLabsClient.class);
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elevenLabsClient.handleTextToVoice(content.trim(), resultPathUrl);
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log.info("3、开始进行AI回答时间:{}",System.currentTimeMillis()/1000);
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//持续返回数据流给客户端
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try {
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String resultOutPathUrl = RuoYiConfig.getProfile() + VOICE_STORAGE_RESULT_DIR + "110_"+resultFileName;
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handleVoice(resultPathUrl,resultOutPathUrl);
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//文件转换成文件流
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ByteBuffer outByteBuffer = convertFileToByteBuffer(resultOutPathUrl);
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//发送文件流数据
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session.getBasicRemote().sendBinary(outByteBuffer);
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// 发送响应确认
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log.info("4、开始进行AI回答时间:{}",System.currentTimeMillis()/1000);
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} catch (IOException e) {
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e.printStackTrace();
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}
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}
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@Override
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public void onComplete() {
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try {
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Map<String,String> resultEntity = new HashMap<>();
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resultEntity.put("msg","done");
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//发送通知告诉客户端已经回答结束了
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session.getBasicRemote().sendText(JSONUtil.toJsonStr(resultEntity));
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} catch (Exception e) {
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throw new RuntimeException(e);
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}
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log.info("5、结束进行AI回答时间:{}",System.currentTimeMillis()/1000);
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}
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@Override
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public void onError(Throwable throwable) {
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throwable.printStackTrace();
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}
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});
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}
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}
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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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// 接收二进制消息（流数据）
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@OnMessage
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public void onBinaryMessage(Session session, @PathParam("clientId") String clientId, ByteBuffer byteBuffer) {
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log.info("1、开始接收数据流时间:{}",System.currentTimeMillis()/1000);
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log.info("客户端ID为:{}", clientId);
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// 处理二进制流数据
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byte[] bytes = new byte[byteBuffer.remaining()];
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//从缓冲区中读取数据并存储到指定的字节数组中
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byteBuffer.get(bytes);
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log.info("2、开始接收数据流时间:{}",System.currentTimeMillis()/1000);
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// 生成唯一文件名
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// String fileName = clientId + "_" + System.currentTimeMillis() + ".webm";
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// String pathUrl = RuoYiConfig.getProfile()+VOICE_STORAGE_DIR + fileName;
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// log.info("文件路径为:{}", pathUrl);
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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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//发送消息
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WebSocket webSocket = cacheWebSocket.get(clientId);
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log.info("获取的socket对象为:{}",webSocket);
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if(webSocket != null){
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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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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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//进行转换
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// 转换音频格式
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AudioFormat format = new AudioFormat(SAMPLE_RATE, BITS_PER_SAMPLE, 1, true, false);
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byte[] outputAudioBytes = convertAudio(bytes, format);
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String base64Audio = Base64.getEncoder().encodeToString(outputAudioBytes);
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String message = "{ \"type\": \"input_audio_buffer.append\", \"audio\": \"" + base64Audio + "\" }";
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webSocket.send(message);
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log.info("4、开始接收数据流时间:{}",System.currentTimeMillis()/1000);
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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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}catch (Exception e){
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e.printStackTrace();
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}
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}
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// 连接关闭时调用
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@OnClose
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public void onClose(Session session, CloseReason reason) {
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System.out.println("WebSocket连接已关闭: " + session.getId() + ", 原因: " + reason.getReasonPhrase());
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}
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// 发生错误时调用
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@OnError
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public void onError(Session session, Throwable throwable) {
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System.err.println("WebSocket错误发生: " + throwable.getMessage());
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throwable.printStackTrace();
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}
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public static byte[] convertAudio(byte[] inputAudioBytes, AudioFormat targetFormat) throws Exception {
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// 将 byte[] 转换为 AudioInputStream
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ByteArrayInputStream byteArrayInputStream = new ByteArrayInputStream(inputAudioBytes);
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AudioInputStream inputAudioStream = new AudioInputStream(byteArrayInputStream, targetFormat, inputAudioBytes.length);
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// 创建目标格式的 AudioInputStream
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AudioInputStream outputAudioStream = AudioSystem.getAudioInputStream(targetFormat, inputAudioStream);
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// 获取输出音频的 byte[]
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ByteArrayOutputStream byteArrayOutputStream = new ByteArrayOutputStream();
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byte[] buffer = new byte[1024];
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int bytesRead;
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// 从 AudioInputStream 读取数据并写入 ByteArrayOutputStream
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while ((bytesRead = outputAudioStream.read(buffer)) != -1) {
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byteArrayOutputStream.write(buffer, 0, bytesRead);
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}
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// 返回转换后的 byte[]
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return byteArrayOutputStream.toByteArray();
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}
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/**
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* 将字节数组保存为WebM文件
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*
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* @param byteData 包含WebM数据的字节数组
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* @param filePath 目标文件路径
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* @return 操作是否成功
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*/
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private boolean saveAsWebM(byte[] byteData, String filePath) {
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// 检查输入参数
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if (byteData == null || byteData.length == 0) {
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System.err.println("字节数组为空，无法生成WebM文件");
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return false;
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}
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if (filePath == null || filePath.trim().isEmpty()) {
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System.err.println("文件路径不能为空");
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return false;
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}
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// 确保文件以.webm结尾
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if (!filePath.toLowerCase().endsWith(".webm")) {
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filePath += ".webm";
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}
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FileOutputStream fos = null;
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try {
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fos = new FileOutputStream(filePath);
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fos.write(byteData);
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fos.flush();
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System.out.println("WebM文件已成功生成: " + filePath);
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return true;
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} catch (IOException e) {
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System.err.println("写入文件时发生错误: " + e.getMessage());
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e.printStackTrace();
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} finally {
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if (fos != null) {
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try {
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fos.close();
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} catch (IOException e) {
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e.printStackTrace();
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}
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}
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}
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return false;
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}
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/**
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* File 转换成 ByteBuffer
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*
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* @param fileUrl 文件路径
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* @return
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*/
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private ByteBuffer convertFileToByteBuffer(String fileUrl) {
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File file = new File(fileUrl);
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try {
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return ByteBuffer.wrap(FileUtils.readFileToByteArray(file));
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} catch (Exception e) {
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e.printStackTrace();
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}
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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", MODEL);
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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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private void handleVoice(String inputPath,String outputPath){
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double trimMs = 270; // 要去掉的尾部时长（毫秒）
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try {
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// 1. 解析音频格式和总长度
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AudioInputStream audioIn = AudioSystem.getAudioInputStream(new File(inputPath));
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AudioFormat format = audioIn.getFormat();
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long totalBytes = audioIn.getFrameLength() * format.getFrameSize(); // 总字节数
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// 2. 计算300毫秒对应的字节数
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float sampleRate = format.getSampleRate(); // 采样率（Hz）
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int frameSize = format.getFrameSize(); // 每帧字节数（位深/8 * 声道数）
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double trimSeconds = trimMs / 1000.0; // 转换为秒
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long trimBytes = (long) (sampleRate * trimSeconds * frameSize); // 要去掉的字节数
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// 3. 计算需要保留的字节数（避免负数）
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long keepBytes = Math.max(0, totalBytes - trimBytes);
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if (keepBytes == 0) {
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System.out.println("音频长度小于300毫秒，无法截断");
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return;
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}
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// 4. 读取并保留前半部分（去掉最后300毫秒）
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try (InputStream in = new FileInputStream(inputPath);
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OutputStream out = new FileOutputStream(outputPath)) {
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byte[] buffer = new byte[4096];
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long totalRead = 0;
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int bytesRead;
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while (totalRead < keepBytes && (bytesRead = in.read(buffer)) != -1) {
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long remaining = keepBytes - totalRead;
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int writeBytes = (remaining < bytesRead) ? (int) remaining : bytesRead;
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out.write(buffer, 0, writeBytes);
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totalRead += writeBytes;
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}
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System.out.println("处理完成，去掉了最后" + trimMs + "毫秒，保留了" + totalRead + "字节");
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}
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} catch (UnsupportedAudioFileException | IOException e) {
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e.printStackTrace();
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}
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}
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||
}
|
||
|