Guía de conexión WebSocket

​

Endpoint de servicio

​

Encabezados de solicitud

​

Flujo de interacción

• Modo ServerCommit: El servidor determina la segmentación del texto y el momento de la síntesis automáticamente.
• Modo Commit: El cliente controla cuándo enviar cada segmento de texto. Llame a input_text_buffer.commit explícitamente para activar la síntesis.

​

Detalles del modo

• En el modo ServerCommit, llame a input_text_buffer.append para almacenar texto en el búfer. El sistema determina el momento de la síntesis automáticamente.
• Llamar a input_text_buffer.commit en el modo ServerCommit sintetiza el búfer actual inmediatamente. Luego, la sesión reanuda el modo ServerCommit.
• En el modo Commit, input_text_buffer.append por sí solo no activa la síntesis. Llame a input_text_buffer.commit para iniciar la síntesis.

1. Conexión: El cliente abre una conexión WebSocket. El servidor devuelve session.created cuando está listo.
2. Configuración de sesión: El cliente envía session.update para configurar la voz, el formato y el modo.
3. Entrada de texto: El cliente envía input_text_buffer.append para agregar texto al búfer.
4. Activador de síntesis:
   • En el modo ServerCommit, la síntesis comienza automáticamente. El cliente puede llamar a input_text_buffer.commit para forzar la síntesis inmediata.
   • En el modo Commit, solo input_text_buffer.commit activa la síntesis.
5. Generación de audio: El servidor envía response.created y luego transmite audio codificado en base64 como eventos response.audio.delta hasta response.audio.done.
6. Finalización de sesión: El cliente envía session.finish para liberar recursos. El servidor devuelve session.finished y cierra la conexión.

{
    "event_id": "event_xxx",
    "type": "session.created",
    "session": {
        "object": "realtime.session",
        "mode": "server_commit",
        "model": "qwen3-tts-flash-realtime",
        "voice": "Cherry",
        "response_format": "pcm",
        "sample_rate": 24000,
        "id": "sess_xxx"
    }
}

​

Ejemplos de código

<dependencies>
    <dependency>
        <groupId>org.java-websocket</groupId>
        <artifactId>Java-WebSocket</artifactId>
        <version>1.5.3</version>
    </dependency>
    <dependency>
        <groupId>com.google.code.gson</groupId>
        <artifactId>gson</artifactId>
        <version>2.10.1</version>
    </dependency>
</dependencies>

1. Crear el cliente
   • Java
   • Python

   Cree un archivo Java llamado TTSRealtimeClient.java y copie el siguiente código en el archivo:

   Copy

   import com.google.gson.Gson;
   import com.google.gson.JsonObject;
   import org.java_websocket.client.WebSocketClient;
   import org.java_websocket.handshake.ServerHandshake;
   
   import java.net.URI;
   import java.util.Base64;
   import java.util.HashMap;
   import java.util.Map;
   import java.util.concurrent.CountDownLatch;
   import java.util.function.Consumer;
   
   /**
    * A client for interacting with the TTS Realtime API.
    *
    * This class provides methods for connecting to the TTS Realtime API, sending text data, retrieving audio output, and managing WebSocket connections.
    */
   public class TTSRealtimeClient {
   
       public enum SessionMode {
           SERVER_COMMIT("server_commit"),
           COMMIT("commit");
           private final String value;
           SessionMode(String value) { this.value = value; }
           public String getValue() { return value; }
       }
   
       /**
        * Audio callback interface
        */
       public interface AudioCallback {
           void onAudio(byte[] audioData);
       }
   
       private final String baseUrl;
       private final String apiKey;
       private final String voice;
       private final SessionMode mode;
       private final String languageType;
       private final AudioCallback audioCallback;
       private final Gson gson = new Gson();
   
       private WebSocketClient ws;
       private CountDownLatch responseDoneLatch;
       private CountDownLatch sessionFinishedLatch;
   
       public TTSRealtimeClient(String baseUrl, String apiKey, String voice,
                                SessionMode mode, AudioCallback audioCallback,
                                String languageType) {
           this.baseUrl = baseUrl;
           this.apiKey = apiKey;
           this.voice = voice;
           this.mode = mode;
           this.audioCallback = audioCallback;
           this.languageType = languageType;
       }
   
       public TTSRealtimeClient(String baseUrl, String apiKey, String voice,
                                SessionMode mode, AudioCallback audioCallback) {
           this(baseUrl, apiKey, voice, mode, audioCallback, "Auto");
       }
   
       /**
        * Establish a WebSocket connection to the TTS Realtime API.
        */
       public void connect() throws Exception {
           Map<String, String> headers = new HashMap<>();
           headers.put("Authorization", "Bearer " + apiKey);
   
           responseDoneLatch = new CountDownLatch(0);
           sessionFinishedLatch = new CountDownLatch(1);
   
           ws = new WebSocketClient(new URI(baseUrl), headers) {
               @Override
               public void onOpen(ServerHandshake handshake) {
                   System.out.println("WebSocket connection established");
                   // Send default session configuration
                   JsonObject session = new JsonObject();
                   session.addProperty("mode", mode.getValue());
                   session.addProperty("voice", TTSRealtimeClient.this.voice);
                   session.addProperty("language_type", languageType);
                   session.addProperty("response_format", "pcm");
                   session.addProperty("sample_rate", 24000);
                   updateSession(session);
               }
   
               @Override
               public void onMessage(String message) {
                   JsonObject event = gson.fromJson(message, JsonObject.class);
                   String eventType = event.has("type") ? event.get("type").getAsString() : "";
   
                   if (!"response.audio.delta".equals(eventType)) {
                       System.out.println("Received event: " + eventType);
                   }
   
                   switch (eventType) {
                       case "error":
                           System.err.println("Error: " + event.get("error"));
                           break;
                       case "session.created":
                           System.out.println("Session created, ID: " +
                               event.getAsJsonObject("session").get("id").getAsString());
                           break;
                       case "session.updated":
                           System.out.println("Session updated, ID: " +
                               event.getAsJsonObject("session").get("id").getAsString());
                           break;
                       case "input_text_buffer.committed":
                           System.out.println("Text buffer committed, item ID: " + event.get("item_id"));
                           break;
                       case "input_text_buffer.cleared":
                           System.out.println("Text buffer cleared");
                           break;
                       case "response.created":
                           System.out.println("Response created, ID: " +
                               event.getAsJsonObject("response").get("id").getAsString());
                           responseDoneLatch = new CountDownLatch(1);
                           break;
                       case "response.output_item.added":
                           System.out.println("Output item added, ID: " +
                               event.getAsJsonObject("item").get("id").getAsString());
                           break;
                       case "response.audio.delta":
                           if (audioCallback != null) {
                               byte[] audioBytes = Base64.getDecoder().decode(
                                   event.get("delta").getAsString());
                               audioCallback.onAudio(audioBytes);
                           }
                           break;
                       case "response.audio.done":
                           System.out.println("Audio generation completed");
                           break;
                       case "response.done":
                           System.out.println("Response completed");
                           responseDoneLatch.countDown();
                           break;
                       case "session.finished":
                           System.out.println("Session finished");
                           sessionFinishedLatch.countDown();
                           break;
                   }
               }
   
               @Override
               public void onClose(int code, String reason, boolean remote) {
                   System.out.println("Connection closed: " + reason);
               }
   
               @Override
               public void onError(Exception ex) {
                   System.err.println("WebSocket error: " + ex.getMessage());
               }
           };
           ws.connectBlocking();
       }
   
       /**
        * Send an event to the server.
        */
       public void sendEvent(JsonObject event) {
           String eventId = "event_" + java.util.UUID.randomUUID().toString();
           event.addProperty("event_id", eventId);
           System.out.println("Sending event: type=" + event.get("type").getAsString()
               + ", event_id=" + eventId);
           ws.send(gson.toJson(event));
       }
   
       /**
        * Update the session configuration.
        */
       public void updateSession(JsonObject config) {
           JsonObject event = new JsonObject();
           event.addProperty("type", "session.update");
           event.add("session", config);
           System.out.println("Updating session configuration: " + event);
           sendEvent(event);
       }
   
       /**
        * Send text data to the API.
        */
       public void appendText(String text) {
           JsonObject event = new JsonObject();
           event.addProperty("type", "input_text_buffer.append");
           event.addProperty("text", text);
           sendEvent(event);
       }
   
       /**
        * Commit the text buffer to trigger processing.
        */
       public void commitTextBuffer() {
           JsonObject event = new JsonObject();
           event.addProperty("type", "input_text_buffer.commit");
           sendEvent(event);
       }
   
       /**
        * Clear the text buffer.
        */
       public void clearTextBuffer() {
           JsonObject event = new JsonObject();
           event.addProperty("type", "input_text_buffer.clear");
           sendEvent(event);
       }
   
       /**
        * End the session.
        */
       public void finishSession() {
           JsonObject event = new JsonObject();
           event.addProperty("type", "session.finish");
           sendEvent(event);
       }
   
       /**
        * Wait for the response.done event.
        */
       public void waitForResponseDone() throws InterruptedException {
           responseDoneLatch.await();
       }
   
       /**
        * Wait for the session.finished event.
        */
       public void waitForSessionFinished() throws InterruptedException {
           sessionFinishedLatch.await();
       }
   
       /**
        * Close the WebSocket connection.
        */
       public void close() {
           if (ws != null) {
               ws.close();
           }
       }
   }
   

   Cree un archivo Python llamado tts_realtime_client.py y copie el siguiente código en el archivo:

   Copy

   # -- coding: utf-8 --
   
   import asyncio
   import websockets
   import json
   import base64
   import time
   import uuid
   from typing import Optional, Callable, Dict, Any
   from enum import Enum
   
   class SessionMode(Enum):
       SERVER_COMMIT = "server_commit"
       COMMIT = "commit"
   
   class TTSRealtimeClient:
       """
       Client for interacting with the TTS Realtime API.
   
       This class provides methods for connecting to the TTS Realtime API, sending text data,
       receiving audio output, and managing WebSocket connections.
   
       Attributes:
           base_url (str):
               Base URL of the Realtime API.
           api_key (str):
               API Key for authentication.
           voice (str):
               Voice used for server-side speech synthesis.
           mode (SessionMode):
               Session mode, either server_commit or commit.
           audio_callback (Callable[[bytes], None]):
               Callback function for receiving audio data.
           language_type(str)
               Language for synthesized speech. Options: Chinese, English, German, Italian, Portuguese, Spanish, Japanese, Korean, French, Russian, Auto
       """
   
       def __init__(
               self,
               base_url: str,
               api_key: str,
               voice: str = "Cherry",
               mode: SessionMode = SessionMode.SERVER_COMMIT,
               audio_callback: Optional[Callable[[bytes], None]] = None,
           language_type: str = "Auto"):
           self.base_url = base_url
           self.api_key = api_key
           self.voice = voice
           self.mode = mode
           self.ws = None
           self.audio_callback = audio_callback
           self.language_type = language_type
   
           # Current response state
           self._current_response_id = None
           self._current_item_id = None
           self._is_responding = False
           self._response_done_future = None
   
       async def connect(self) -> None:
           """Establish WebSocket connection with the TTS Realtime API."""
           headers = {
               "Authorization": f"Bearer {self.api_key}"
           }
   
           self.ws = await websockets.connect(self.base_url, additional_headers=headers)
   
           # Set default session configuration
           await self.update_session({
               "mode": self.mode.value,
               "voice": self.voice,
               "language_type": self.language_type,
               "response_format": "pcm",
               "sample_rate": 24000
           })
   
       async def send_event(self, event) -> None:
           """Send an event to the server."""
           event['event_id'] = "event_" + uuid.uuid4().hex
           print(f"Sending event: type={event['type']}, event_id={event['event_id']}")
           await self.ws.send(json.dumps(event))
   
       async def update_session(self, config: Dict[str, Any]) -> None:
           """Update session configuration."""
           event = {
               "type": "session.update",
               "session": config
           }
           print("Updating session configuration: ", event)
           await self.send_event(event)
   
       async def append_text(self, text: str) -> None:
           """Send text data to the API."""
           event = {
               "type": "input_text_buffer.append",
               "text": text
           }
           await self.send_event(event)
   
       async def commit_text_buffer(self) -> None:
           """Commit text buffer to trigger processing."""
           event = {
               "type": "input_text_buffer.commit"
           }
           await self.send_event(event)
   
       async def clear_text_buffer(self) -> None:
           """Clear the text buffer."""
           event = {
               "type": "input_text_buffer.clear"
           }
           await self.send_event(event)
   
       async def finish_session(self) -> None:
           """End the session."""
           event = {
               "type": "session.finish"
           }
           await self.send_event(event)
   
       async def wait_for_response_done(self):
           """Wait for the response.done event"""
           if self._response_done_future:
               await self._response_done_future
   
       async def handle_messages(self) -> None:
           """Handle messages from the server."""
           try:
               async for message in self.ws:
                   event = json.loads(message)
                   event_type = event.get("type")
   
                   if event_type != "response.audio.delta":
                       print(f"Received event: {event_type}")
   
                   if event_type == "error":
                       print("Error: ", event.get('error', {}))
                       continue
                   elif event_type == "session.created":
                       print("Session created, ID: ", event.get('session', {}).get('id'))
                   elif event_type == "session.updated":
                       print("Session updated, ID: ", event.get('session', {}).get('id'))
                   elif event_type == "input_text_buffer.committed":
                       print("Text buffer committed, item ID: ", event.get('item_id'))
                   elif event_type == "input_text_buffer.cleared":
                       print("Text buffer cleared")
                   elif event_type == "response.created":
                       self._current_response_id = event.get("response", {}).get("id")
                       self._is_responding = True
                       # Create a new future to wait for response.done
                       self._response_done_future = asyncio.Future()
                       print("Response created, ID: ", self._current_response_id)
                   elif event_type == "response.output_item.added":
                       self._current_item_id = event.get("item", {}).get("id")
                       print("Output item added, ID: ", self._current_item_id)
                   # Handle audio delta
                   elif event_type == "response.audio.delta" and self.audio_callback:
                       audio_bytes = base64.b64decode(event.get("delta", ""))
                       self.audio_callback(audio_bytes)
                   elif event_type == "response.audio.done":
                       print("Audio generation completed")
                   elif event_type == "response.done":
                       self._is_responding = False
                       self._current_response_id = None
                       self._current_item_id = None
                       # Mark future as done
                       if self._response_done_future and not self._response_done_future.done():
                           self._response_done_future.set_result(True)
                       print("Response completed")
                   elif event_type == "session.finished":
                       print("Session finished")
   
           except websockets.exceptions.ConnectionClosed:
               print("Connection closed")
           except Exception as e:
               print("Error handling messages: ", str(e))
   
       async def close(self) -> None:
           """Close the WebSocket connection."""
           if self.ws:
               await self.ws.close()
   

2. Seleccionar un modo de síntesis de voz La API Realtime admite dos modos:
   • Modo server_commit El servidor gestiona automáticamente la segmentación de texto y los tiempos de síntesis. El cliente solo envía texto. Adecuado para escenarios de baja latencia (como navegación GPS).
   • modo commit El cliente agrega texto a un búfer y activa explícitamente la síntesis. Adecuado para escenarios que requieren un control preciso de la segmentación de oraciones (como transmisiones de noticias).
   • modo server_commit
   • modo de confirmación

   • Java
   • Python

   En el mismo directorio que TTSRealtimeClient.java, cree otro archivo Java llamado ServerCommit.java y copie el siguiente código en el archivo:

   Copy

   import javax.sound.sampled.*;
   import java.io.*;
   import java.util.ArrayList;
   import java.util.List;
   import java.util.concurrent.ConcurrentLinkedQueue;
   import java.util.concurrent.atomic.AtomicBoolean;
   
   public class ServerCommit {
       // The following is the configuration for the Singapore region.
       private static final String URL = "wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime";
       // The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/model-studio/get-api-key
       // If you have not configured the environment variable, replace the next line with your Model Studio API Key: private static final String API_KEY = "sk-xxx";
       private static final String API_KEY = System.getenv("DASHSCOPE_API_KEY");
       private static final int SAMPLE_RATE = 24000;
   
       // Audio data cache
       private static final List<byte[]> audioChunks = new ArrayList<>();
       // Real-time playback queue
       private static final ConcurrentLinkedQueue<byte[]> playbackQueue = new ConcurrentLinkedQueue<>();
       private static final AtomicBoolean playing = new AtomicBoolean(true);
   
       public static void main(String[] args) throws Exception {
           if (API_KEY == null || API_KEY.isEmpty()) {
               throw new IllegalStateException("Please set the DASHSCOPE_API_KEY environment variable");
           }
   
           // Initialize audio playback
           AudioFormat format = new AudioFormat(SAMPLE_RATE, 16, 1, true, false);
           DataLine.Info info = new DataLine.Info(SourceDataLine.class, format);
           SourceDataLine audioLine = (SourceDataLine) AudioSystem.getLine(info);
           audioLine.open(format);
           audioLine.start();
   
           // Start the playback thread
           Thread playerThread = new Thread(() -> {
               while (playing.get() || !playbackQueue.isEmpty()) {
                   byte[] chunk = playbackQueue.poll();
                   if (chunk != null) {
                       audioLine.write(chunk, 0, chunk.length);
                   } else {
                       try { Thread.sleep(10); } catch (InterruptedException ignored) {}
                   }
               }
           });
           playerThread.start();
   
           // Create the TTS client
           TTSRealtimeClient client = new TTSRealtimeClient(
               URL, API_KEY, "Cherry",
               TTSRealtimeClient.SessionMode.SERVER_COMMIT,
               audioData -> {
                   playbackQueue.add(audioData);
                   audioChunks.add(audioData);
                   System.out.println("Received audio data: " + audioData.length + " bytes");
               }
           );
   
           client.connect();
   
           // Send text fragments
           String[] textFragments = {
               "Alibaba Cloud's large language model platform, Model Studio, is an all-in-one platform for developing and building large language model applications.",
               "Both developers and business users can deeply participate in the design and development of large language model applications.",
               "You can develop a large language model application in five minutes using a simple interface,",
               "or train a dedicated model in a few hours, allowing you to focus more energy on application innovation."
           };
   
           System.out.println("Starting to send text...");
           for (String text : textFragments) {
               System.out.println("Sending fragment: " + text);
               client.appendText(text);
               Thread.sleep(100);
           }
   
           Thread.sleep(1000);
           client.finishSession();
   
           // Wait for the response to complete
           client.waitForResponseDone();
           client.waitForSessionFinished();
           client.close();
   
           // Wait for playback to complete
           playing.set(false);
           playerThread.join();
           audioLine.drain();
           audioLine.close();
   
           // Save the audio file
           saveWav("output.wav");
           System.out.println("Done");
       }
   
       private static void saveWav(String filename) throws IOException {
           if (audioChunks.isEmpty()) {
               System.out.println("No audio data to save");
               return;
           }
           ByteArrayOutputStream bos = new ByteArrayOutputStream();
           for (byte[] chunk : audioChunks) {
               bos.write(chunk);
           }
           byte[] allAudio = bos.toByteArray();
           AudioFormat format = new AudioFormat(SAMPLE_RATE, 16, 1, true, false);
           AudioInputStream ais = new AudioInputStream(
               new ByteArrayInputStream(allAudio), format, allAudio.length / 2);
           new File("outputs").mkdirs();
           AudioSystem.write(ais, AudioFileFormat.Type.WAVE,
               new File("outputs/" + filename));
           System.out.println("Audio saved to: outputs/" + filename);
       }
   }
   

   Compile y ejecute ServerCommit.java para escuchar el audio generado por Realtime API en tiempo real.

   En el mismo directorio que tts_realtime_client.py, cree otro archivo Python llamado server_commit.py y copie el siguiente código en el archivo:

   Copy

   import os
   import asyncio
   import logging
   import wave
   from tts_realtime_client import TTSRealtimeClient, SessionMode
   import pyaudio
   
   # QwenTTS service configuration
   # The following is the configuration for the Singapore region.
   URL = "wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime"
   # The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/model-studio/get-api-key
   # If you have not configured the environment variable, replace the next line with your Model Studio API Key: API_KEY="sk-xxx"
   API_KEY = os.getenv("DASHSCOPE_API_KEY")
   
   if not API_KEY:
       raise ValueError("Please set DASHSCOPE_API_KEY environment variable")
   
   # Collect audio data
   _audio_chunks = []
   # Real-time playback related
   _AUDIO_SAMPLE_RATE = 24000
   _audio_pyaudio = pyaudio.PyAudio()
   _audio_stream = None  # Will be opened at runtime
   
   def _audio_callback(audio_bytes: bytes):
       """TTSRealtimeClient audio callback: real-time playback and caching"""
       global _audio_stream
       if _audio_stream is not None:
           try:
               _audio_stream.write(audio_bytes)
           except Exception as exc:
               logging.error(f"PyAudio playback error: {exc}")
       _audio_chunks.append(audio_bytes)
       logging.info(f"Received audio chunk: {len(audio_bytes)} bytes")
   
   def _save_audio_to_file(filename: str = "output.wav", sample_rate: int = 24000) -> bool:
       """Save collected audio data as a WAV file"""
       if not _audio_chunks:
           logging.warning("No audio data to save")
           return False
   
       try:
           audio_data = b"".join(_audio_chunks)
           with wave.open(filename, 'wb') as wav_file:
               wav_file.setnchannels(1)  # Mono
               wav_file.setsampwidth(2)  # 16-bit
               wav_file.setframerate(sample_rate)
               wav_file.writeframes(audio_data)
           logging.info(f"Audio saved to: {filename}")
           return True
       except Exception as exc:
           logging.error(f"Failed to save audio: {exc}")
           return False
   
   async def _produce_text(client: TTSRealtimeClient):
       """Send text fragments to the server"""
       text_fragments = [
           "Alibaba Cloud's large language model platform, Model Studio, is an all-in-one platform for developing and building large language model applications.",
           "Both developers and business users can deeply participate in the design and development of large language model applications.",
           "You can develop a large language model application in five minutes using a simple interface,",
           "or train a dedicated model in a few hours, allowing you to focus more energy on application innovation.",
       ]
   
       logging.info("Sending text fragments…")
       for text in text_fragments:
           logging.info(f"Sending fragment: {text}")
           await client.append_text(text)
           await asyncio.sleep(0.1)  # Brief delay between fragments
   
       # Wait for the server to finish internal processing before ending the session
       await asyncio.sleep(1.0)
       await client.finish_session()
   
   async def _run_demo():
       """Run the complete demo"""
       global _audio_stream
       # Open PyAudio output stream
       _audio_stream = _audio_pyaudio.open(
           format=pyaudio.paInt16,
           channels=1,
           rate=_AUDIO_SAMPLE_RATE,
           output=True,
           frames_per_buffer=1024
       )
   
       client = TTSRealtimeClient(
           base_url=URL,
           api_key=API_KEY,
           voice="Cherry",
           mode=SessionMode.SERVER_COMMIT,
           audio_callback=_audio_callback
       )
   
       # Establish connection
       await client.connect()
   
       # Run message handling and text sending in parallel
       consumer_task = asyncio.create_task(client.handle_messages())
       producer_task = asyncio.create_task(_produce_text(client))
   
       await producer_task  # Wait for text sending to complete
   
       # Wait for response.done
       await client.wait_for_response_done()
   
       # Close connection and cancel consumer task
       await client.close()
       consumer_task.cancel()
   
       # Close audio stream
       if _audio_stream is not None:
           _audio_stream.stop_stream()
           _audio_stream.close()
       _audio_pyaudio.terminate()
   
       # Save audio data
       os.makedirs("outputs", exist_ok=True)
       _save_audio_to_file(os.path.join("outputs", "qwen_tts_output.wav"))
   
   def main():
       """Synchronous entry point"""
       logging.basicConfig(
           level=logging.INFO,
           format='%(asctime)s [%(levelname)s] %(message)s',
           datefmt='%Y-%m-%d %H:%M:%S'
       )
       logging.info("Starting QwenTTS Realtime Client demo…")
       asyncio.run(_run_demo())
   
   if __name__ == "__main__":
       main()
   

   Ejecute server_commit.py para escuchar en tiempo real el audio generado por Realtime API.

   • Java
   • Python

   En el mismo directorio que TTSRealtimeClient.java, cree otro archivo Java llamado Commit.java y copie el siguiente código en el archivo:

   Copy

   import javax.sound.sampled.*;
   import java.io.*;
   import java.util.ArrayList;
   import java.util.List;
   import java.util.Scanner;
   import java.util.concurrent.ConcurrentLinkedQueue;
   import java.util.concurrent.atomic.AtomicBoolean;
   
   public class Commit {
       // The following is the configuration for the Singapore region.
       private static final String URL = "wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime";
       // The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/model-studio/get-api-key
       // If you have not configured the environment variable, replace the next line with your Model Studio API Key: private static final String API_KEY = "sk-xxx";
       private static final String API_KEY = System.getenv("DASHSCOPE_API_KEY");
       private static final int SAMPLE_RATE = 24000;
   
       private static final List<byte[]> audioChunks = new ArrayList<>();
       private static final ConcurrentLinkedQueue<byte[]> playbackQueue = new ConcurrentLinkedQueue<>();
       private static final AtomicBoolean playing = new AtomicBoolean(true);
   
       public static void main(String[] args) throws Exception {
           if (API_KEY == null || API_KEY.isEmpty()) {
               throw new IllegalStateException("Please set the DASHSCOPE_API_KEY environment variable");
           }
   
           // Initialize audio playback
           AudioFormat format = new AudioFormat(SAMPLE_RATE, 16, 1, true, false);
           DataLine.Info info = new DataLine.Info(SourceDataLine.class, format);
           SourceDataLine audioLine = (SourceDataLine) AudioSystem.getLine(info);
           audioLine.open(format);
           audioLine.start();
   
           // Start the playback thread
           Thread playerThread = new Thread(() -> {
               while (playing.get() || !playbackQueue.isEmpty()) {
                   byte[] chunk = playbackQueue.poll();
                   if (chunk != null) {
                       audioLine.write(chunk, 0, chunk.length);
                   } else {
                       try { Thread.sleep(10); } catch (InterruptedException ignored) {}
                   }
               }
           });
           playerThread.start();
   
           // Create a TTS client (commit mode)
           TTSRealtimeClient client = new TTSRealtimeClient(
               URL, API_KEY, "Cherry",
               TTSRealtimeClient.SessionMode.COMMIT,
               audioData -> {
                   playbackQueue.add(audioData);
                   audioChunks.add(audioData);
                   System.out.println("Received audio data: " + audioData.length + " bytes");
               }
           );
   
           client.connect();
   
           // Interactive input
           System.out.println("Enter text (press Enter directly to send a commit event and end the session, press Ctrl+D to exit the program):");
           Scanner scanner = new Scanner(System.in);
           while (true) {
               System.out.print("> ");
               if (!scanner.hasNextLine()) {
                   client.finishSession();
                   break;
               }
               String userText = scanner.nextLine();
               if (userText.isEmpty()) {
                   // Empty input: commit the buffer and end the session
                   System.out.println("Empty input, sending commit event and ending the session");
                   client.commitTextBuffer();
                   Thread.sleep(300);
                   client.finishSession();
                   break;
               } else {
                   System.out.println("Sending text: " + userText);
                   client.appendText(userText);
               }
           }
           scanner.close();
   
           // Wait for the response to complete
           client.waitForResponseDone();
           client.waitForSessionFinished();
           client.close();
   
           // Wait for playback to complete
           playing.set(false);
           playerThread.join();
           audioLine.drain();
           audioLine.close();
   
           // Save the audio file
           saveWav("output.wav");
           System.out.println("Done");
       }
   
       private static void saveWav(String filename) throws IOException {
           if (audioChunks.isEmpty()) {
               System.out.println("No audio data to save");
               return;
           }
           ByteArrayOutputStream bos = new ByteArrayOutputStream();
           for (byte[] chunk : audioChunks) {
               bos.write(chunk);
           }
           byte[] allAudio = bos.toByteArray();
           AudioFormat format = new AudioFormat(SAMPLE_RATE, 16, 1, true, false);
           AudioInputStream ais = new AudioInputStream(
               new ByteArrayInputStream(allAudio), format, allAudio.length / 2);
           new File("outputs").mkdirs();
           AudioSystem.write(ais, AudioFileFormat.Type.WAVE,
               new File("outputs/" + filename));
           System.out.println("Audio saved to: outputs/" + filename);
       }
   }
   

   Compile y ejecute Commit.java. Puede ingresar texto para sintetizar varias veces. Presione Enter sin ingresar texto para escuchar el audio devuelto por Realtime API a través del altavoz.

   En el mismo directorio que tts_realtime_client.py, cree otro archivo Python llamado commit.py y copie el siguiente código en el archivo:

   Copy

   import os
   import asyncio
   import logging
   import wave
   from tts_realtime_client import TTSRealtimeClient, SessionMode
   import pyaudio
   
   # QwenTTS service configuration
   # The following is the configuration for the Singapore region.
   URL = "wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime"
   # The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/model-studio/get-api-key
   # If you have not configured the environment variable, replace the next line with your Model Studio API Key: API_KEY="sk-xxx"
   API_KEY = os.getenv("DASHSCOPE_API_KEY")
   
   if not API_KEY:
       raise ValueError("Please set DASHSCOPE_API_KEY environment variable")
   
   # Collect audio data
   _audio_chunks = []
   _AUDIO_SAMPLE_RATE = 24000
   _audio_pyaudio = pyaudio.PyAudio()
   _audio_stream = None
   
   def _audio_callback(audio_bytes: bytes):
       """TTSRealtimeClient audio callback: real-time playback and caching"""
       global _audio_stream
       if _audio_stream is not None:
           try:
               _audio_stream.write(audio_bytes)
           except Exception as exc:
               logging.error(f"PyAudio playback error: {exc}")
       _audio_chunks.append(audio_bytes)
       logging.info(f"Received audio chunk: {len(audio_bytes)} bytes")
   
   def _save_audio_to_file(filename: str = "output.wav", sample_rate: int = 24000) -> bool:
       """Save collected audio data as a WAV file"""
       if not _audio_chunks:
           logging.warning("No audio data to save")
           return False
   
       try:
           audio_data = b"".join(_audio_chunks)
           with wave.open(filename, 'wb') as wav_file:
               wav_file.setnchannels(1)  # Mono
               wav_file.setsampwidth(2)  # 16-bit
               wav_file.setframerate(sample_rate)
               wav_file.writeframes(audio_data)
           logging.info(f"Audio saved to: {filename}")
           return True
       except Exception as exc:
           logging.error(f"Failed to save audio: {exc}")
           return False
   
   async def _user_input_loop(client: TTSRealtimeClient):
       """Continuously get user input and send text. When user enters empty text, send a commit event and end the current session"""
       print("Enter text (press Enter directly to send a commit event and end the current session, press Ctrl+C or Ctrl+D to exit the program):")
   
       while True:
           try:
               user_text = input("> ")
               if not user_text:  # User input is empty
                   # Empty input is treated as end of a conversation: commit buffer -> end session -> break loop
                   logging.info("Empty input, sending commit event and ending current session")
                   await client.commit_text_buffer()
                   # Wait briefly for the server to process the commit, preventing premature session end that could lose audio
                   await asyncio.sleep(0.3)
                   await client.finish_session()
                   break  # Exit user input loop directly, no need to press Enter again
               else:
                   logging.info(f"Sending text: {user_text}")
                   await client.append_text(user_text)
   
           except EOFError:  # User pressed Ctrl+D
               break
           except KeyboardInterrupt:  # User pressed Ctrl+C
               break
   
       # End session
       logging.info("Ending session...")
   async def _run_demo():
       """Run the complete demo"""
       global _audio_stream
       # Open PyAudio output stream
       _audio_stream = _audio_pyaudio.open(
           format=pyaudio.paInt16,
           channels=1,
           rate=_AUDIO_SAMPLE_RATE,
           output=True,
           frames_per_buffer=1024
       )
   
       client = TTSRealtimeClient(
           base_url=URL,
           api_key=API_KEY,
           voice="Cherry",
           mode=SessionMode.COMMIT,  # Changed to COMMIT mode
           audio_callback=_audio_callback
       )
   
       # Establish connection
       await client.connect()
   
       # Run message handling and user input in parallel
       consumer_task = asyncio.create_task(client.handle_messages())
       producer_task = asyncio.create_task(_user_input_loop(client))
   
       await producer_task  # Wait for user input to complete
   
       # Wait for response.done
       await client.wait_for_response_done()
   
       # Close connection and cancel consumer task
       await client.close()
       consumer_task.cancel()
   
       # Close audio stream
       if _audio_stream is not None:
           _audio_stream.stop_stream()
           _audio_stream.close()
       _audio_pyaudio.terminate()
   
       # Save audio data
       os.makedirs("outputs", exist_ok=True)
       _save_audio_to_file(os.path.join("outputs", "qwen_tts_output.wav"))
   
   def main():
       logging.basicConfig(
           level=logging.INFO,
           format='%(asctime)s [%(levelname)s] %(message)s',
           datefmt='%Y-%m-%d %H:%M:%S'
       )
       logging.info("Starting QwenTTS Realtime Client demo…")
       asyncio.run(_run_demo())
   
   if __name__ == "__main__":
       main()
   

   Ejecute commit.py. Puede ingresar texto para sintetizar varias veces. Presione Enter sin ingresar texto para escuchar el audio devuelto por Realtime API a través del altavoz.