Alibaba Cloud Model Studio:API WebSocket pour la synthèse vocale en temps réel Qwen-TTS

Connectez-vous au service de synthèse vocale en temps réel Qwen-TTS via WebSocket. Cette rubrique présente l'endpoint du service, les en-têtes de requête et le flux d'interaction.

Pour une présentation des modèles et des conseils de sélection, consultez Synthèse vocale en temps réel.

L'API Realtime de Qwen-TTS repose sur le protocole WebSocket. Les applications Java et Python peuvent utiliser le SDK DashScope pour s'affranchir de la gestion directe du WebSocket. Pour les autres langages, connectez-vous directement à l'aide d'une bibliothèque WebSocket.

Endpoint du service

Utilisez l'URL suivante. Définissez le modèle à l'aide du paramètre de requête model :

En-têtes de requête

Définissez les en-têtes de requête suivants :

Flux d'interaction

Pour une description détaillée des événements client et serveur, reportez-vous aux rubriques Événements client et Événements serveur.

L'API prend en charge deux modes :

• Mode ServerCommit : le serveur gère automatiquement la segmentation du texte et le déclenchement de la synthèse.
• Mode Commit : le client maîtrise le moment de soumission de chaque segment de texte. Appelez explicitement input_text_buffer.commit pour lancer la synthèse.

Détails des modes

• En mode ServerCommit, utilisez input_text_buffer.append pour mettre le texte en mémoire tampon. Le système détermine alors automatiquement le moment opportun pour la synthèse.
• Si vous appelez input_text_buffer.commit en mode ServerCommit, la synthèse du tampon actuel s'effectue immédiatement, puis la session reprend son fonctionnement normal en mode ServerCommit.
• En mode Commit, l'appel seul à input_text_buffer.append ne déclenche pas la synthèse. Appelez input_text_buffer.commit pour démarrer le traitement.

1. Connexion : ouvrez une connexion WebSocket depuis le client. Le serveur renvoie session.created lorsqu'il est prêt.

2. Configuration de la session : envoyez session.update pour configurer la voix, le format audio et le mode de fonctionnement.

3. Saisie du texte : transmettez input_text_buffer.append afin d'ajouter du texte au tampon.

4. Déclenchement de la synthèse :

   • En mode ServerCommit, la synthèse démarre automatiquement. Le client peut toutefois appeler input_text_buffer.commit pour forcer une synthèse immédiate.
   • En mode Commit, seul l'appel à input_text_buffer.commit initie la synthèse.

5. Génération audio : le serveur émet response.created, puis diffuse l'audio encodé en base64 sous forme d'événements response.audio.delta jusqu'à la réception de response.audio.done.

6. Clôture de la session : envoyez session.finish pour libérer les ressources. Le serveur répond par session.finished et ferme la connexion.

Une fois la connexion établie, le serveur retourne l'événement session.created suivant :

{
    "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"
    }
}

Exemples de code

Ces exemples fournissent des clients Python et Java et les deux modes de validation. Définissez DASHSCOPE_API_KEY pour la région de l’adresse utilisée. Python nécessite websockets et pyaudio ; Java nécessite Java-WebSocket et Gson, et utilise Java Sound pour la lecture. Chaque événement client utilise un UUID unique.

<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>

pip install websockets pyaudio

1. Créer le client

   Java

   Créez un fichier Java nommé TTSRealtimeClient.java et copiez-y le code suivant :

   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();
           }
       }
   }
   

   Python

   Créez un fichier Python nommé tts_realtime_client.py et copiez-y le code suivant :

   # -- 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. Choisir un mode de synthèse vocale

   L'API Realtime prend en charge deux modes :

   • Mode server_commit

     Le serveur gère automatiquement la segmentation du texte et le timing de la synthèse. Le client se contente d'envoyer le texte. Ce mode convient aux scénarios à faible latence (comme la navigation GPS).

   • Mode commit

     Le client ajoute le texte dans un tampon et déclenche explicitement la synthèse. Cette approche est recommandée lorsque vous devez contrôler précisément la segmentation des phrases (par exemple pour un journal télévisé).

   Mode server_commit

   Java

   Dans le même répertoire que TTSRealtimeClient.java, créez un autre fichier Java nommé ServerCommit.java et copiez-y le code suivant :

   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);
       }
   }
   

   Compilez et exécutez ServerCommit.java pour écouter en temps réel l'audio généré par l'API Realtime.

   Python

   Dans le même répertoire que tts_realtime_client.py, créez un autre fichier Python nommé server_commit.py et copiez-y le code suivant :

   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()
   

   Exécutez server_commit.py pour écouter en temps réel l'audio généré par l'API Realtime.

   Mode commit

   Java

   Dans le même répertoire que TTSRealtimeClient.java, créez un autre fichier Java nommé Commit.java et copiez-y le code suivant :

   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);
       }
   }
   

   Compilez et exécutez Commit.java. Il est possible de saisir du texte à synthétiser plusieurs fois de suite. Appuyez simplement sur Entrée sans texte pour diffuser l'audio généré par l'API Realtime sur votre haut-parleur.

   Python

   Dans le même répertoire que tts_realtime_client.py, créez un autre fichier Python nommé commit.py et copiez-y le code suivant :

   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()
   

   Exécutez commit.py. Vous avez la possibilité de saisir du texte à synthétiser à plusieurs reprises. Appuyez sur Entrée sans saisir de texte pour écouter via le haut-parleur l'audio renvoyé par l'API Realtime.