Alibaba Cloud Model Studio:Real-time speech synthesis - Qwen

DashScope SDK

import os
import base64
import threading
import time
import dashscope
from dashscope.audio.qwen_tts_realtime import *


qwen_tts_realtime: QwenTtsRealtime = None
text_to_synthesize = [
    'Right? I love supermarkets like this.',
    'Especially during Chinese New Year,',
    'I go shopping at supermarkets.',
    'And I feel',
    'absolutely thrilled!',
    'I want to buy so many things!'
]

DO_VIDEO_TEST = False

def init_dashscope_api_key():
    """
        Set your DashScope API key. More information:
        https://github.com/aliyun/alibabacloud-bailian-speech-demo/blob/master/PREREQUISITES.md
    """

    # API keys differ between the Singapore and Beijing regions. Get an API key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
    if 'DASHSCOPE_API_KEY' in os.environ:
        dashscope.api_key = os.environ[
            'DASHSCOPE_API_KEY']  # Load API key from environment variable DASHSCOPE_API_KEY
    else:
        dashscope.api_key = 'your-dashscope-api-key'  # Set API key manually



class MyCallback(QwenTtsRealtimeCallback):
    def __init__(self):
        self.complete_event = threading.Event()
        self.file = open('result_24k.pcm', 'wb')

    def on_open(self) -> None:
        print('connection opened, init player')

    def on_close(self, close_status_code, close_msg) -> None:
        self.file.close()
        print('connection closed with code: {}, msg: {}, destroy player'.format(close_status_code, close_msg))

    def on_event(self, response: str) -> None:
        try:
            global qwen_tts_realtime
            type = response['type']
            if 'session.created' == type:
                print('start session: {}'.format(response['session']['id']))
            if 'response.audio.delta' == type:
                recv_audio_b64 = response['delta']
                self.file.write(base64.b64decode(recv_audio_b64))
            if 'response.done' == type:
                print(f'response {qwen_tts_realtime.get_last_response_id()} done')
            if 'session.finished' == type:
                print('session finished')
                self.complete_event.set()
        except Exception as e:
            print('[Error] {}'.format(e))
            return

    def wait_for_finished(self):
        self.complete_event.wait()


if __name__  == '__main__':
    init_dashscope_api_key()

    print('Initializing ...')

    callback = MyCallback()

    qwen_tts_realtime = QwenTtsRealtime(
        # To use instruction control, replace the model with qwen3-tts-instruct-flash-realtime
        model='qwen3-tts-flash-realtime',
        callback=callback,
        # This URL is for the Singapore region. If you use the Beijing region, replace it with: wss://dashscope.aliyuncs.com/api-ws/v1/realtime
        url='wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime'
        )

    qwen_tts_realtime.connect()
    qwen_tts_realtime.update_session(
        voice = 'Cherry',
        response_format = AudioFormat.PCM_24000HZ_MONO_16BIT,
        # To use instruction control, uncomment the following lines and replace the model with qwen3-tts-instruct-flash-realtime
        # instructions='Speak quickly with a rising intonation, suitable for introducing fashion products.',
        # optimize_instructions=True,
        mode = 'server_commit'        
    )
    for text_chunk in text_to_synthesize:
        print(f'send text: {text_chunk}')
        qwen_tts_realtime.append_text(text_chunk)
        time.sleep(0.1)
    qwen_tts_realtime.finish()
    callback.wait_for_finished()
    print('[Metric] session: {}, first audio delay: {}'.format(
                    qwen_tts_realtime.get_session_id(), 
                    qwen_tts_realtime.get_first_audio_delay(),
                    ))

import base64
import os
import threading
import dashscope
from dashscope.audio.qwen_tts_realtime import *


qwen_tts_realtime: QwenTtsRealtime = None
text_to_synthesize = [
    'This is the first sentence.',
    'This is the second sentence.',
    'This is the third sentence.',
]

DO_VIDEO_TEST = False

def init_dashscope_api_key():
    """
        Set your DashScope API key. More information:
        https://github.com/aliyun/alibabacloud-bailian-speech-demo/blob/master/PREREQUISITES.md
    """

    # API keys differ between the Singapore and Beijing regions. Get an API key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
    if 'DASHSCOPE_API_KEY' in os.environ:
        dashscope.api_key = os.environ[
            'DASHSCOPE_API_KEY']  # Load API key from environment variable DASHSCOPE_API_KEY
    else:
        dashscope.api_key = 'your-dashscope-api-key'  # Set API key manually



class MyCallback(QwenTtsRealtimeCallback):
    def __init__(self):
        super().__init__()
        self.response_counter = 0
        self.complete_event = threading.Event()
        self.file = open(f'result_{self.response_counter}_24k.pcm', 'wb')

    def reset_event(self):
        self.response_counter += 1
        self.file = open(f'result_{self.response_counter}_24k.pcm', 'wb')
        self.complete_event = threading.Event()

    def on_open(self) -> None:
        print('connection opened, init player')

    def on_close(self, close_status_code, close_msg) -> None:
        print('connection closed with code: {}, msg: {}, destroy player'.format(close_status_code, close_msg))

    def on_event(self, response: str) -> None:
        try:
            global qwen_tts_realtime
            type = response['type']
            if 'session.created' == type:
                print('start session: {}'.format(response['session']['id']))
            if 'response.audio.delta' == type:
                recv_audio_b64 = response['delta']
                self.file.write(base64.b64decode(recv_audio_b64))
            if 'response.done' == type:
                print(f'response {qwen_tts_realtime.get_last_response_id()} done')
                self.complete_event.set()
                self.file.close()
            if 'session.finished' == type:
                print('session finished')
                self.complete_event.set()
        except Exception as e:
            print('[Error] {}'.format(e))
            return

    def wait_for_response_done(self):
        self.complete_event.wait()


if __name__  == '__main__':
    init_dashscope_api_key()

    print('Initializing ...')

    callback = MyCallback()

    qwen_tts_realtime = QwenTtsRealtime(
        # To use instruction control, replace the model with qwen3-tts-instruct-flash-realtime
        model='qwen3-tts-flash-realtime',
        callback=callback, 
        # This URL is for the Singapore region. If you use the Beijing region, replace it with: wss://dashscope.aliyuncs.com/api-ws/v1/realtime
        url='wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime'
        )

    qwen_tts_realtime.connect()
    qwen_tts_realtime.update_session(
        voice = 'Cherry',
        response_format = AudioFormat.PCM_24000HZ_MONO_16BIT,
        # To use instruction control, uncomment the following lines and replace the model with qwen3-tts-instruct-flash-realtime
        # instructions='Speak quickly with a rising intonation, suitable for introducing fashion products.',
        # optimize_instructions=True,
        mode = 'commit'        
    )
    print(f'send text: {text_to_synthesize[0]}')
    qwen_tts_realtime.append_text(text_to_synthesize[0])
    qwen_tts_realtime.commit()
    callback.wait_for_response_done()
    callback.reset_event()
    
    print(f'send text: {text_to_synthesize[1]}')
    qwen_tts_realtime.append_text(text_to_synthesize[1])
    qwen_tts_realtime.commit()
    callback.wait_for_response_done()
    callback.reset_event()

    print(f'send text: {text_to_synthesize[2]}')
    qwen_tts_realtime.append_text(text_to_synthesize[2])
    qwen_tts_realtime.commit()
    callback.wait_for_response_done()
    
    qwen_tts_realtime.finish()
    print('[Metric] session: {}, first audio delay: {}'.format(
                    qwen_tts_realtime.get_session_id(), 
                    qwen_tts_realtime.get_first_audio_delay(),
                    ))

// The Dashscope SDK version must be 2.21.16 or later.
import com.alibaba.dashscope.audio.qwen_tts_realtime.*;
import com.alibaba.dashscope.exception.NoApiKeyException;
import com.google.gson.JsonObject;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.AudioSystem;
import java.io.FileNotFoundException;
import java.io.IOException;
import java.util.Base64;
import java.util.Queue;
import java.util.concurrent.CountDownLatch;
import java.util.concurrent.atomic.AtomicReference;
import java.util.concurrent.ConcurrentLinkedQueue;
import java.util.concurrent.atomic.AtomicBoolean;

public class Main {
    static String[] textToSynthesize = {
            "Right? I especially love this kind of supermarket.",
            "Especially during the New Year.",
            "Going to the supermarket.",
            "It just makes me feel.",
            "Super, super happy!",
            "I want to buy so many things!"
    };

    // Real-time PCM audio player class
    public static class RealtimePcmPlayer {
        private int sampleRate;
        private SourceDataLine line;
        private AudioFormat audioFormat;
        private Thread decoderThread;
        private Thread playerThread;
        private AtomicBoolean stopped = new AtomicBoolean(false);
        private Queue<String> b64AudioBuffer = new ConcurrentLinkedQueue<>();
        private Queue<byte[]> RawAudioBuffer = new ConcurrentLinkedQueue<>();

        // The constructor initializes the audio format and audio line.
        public RealtimePcmPlayer(int sampleRate) throws LineUnavailableException {
            this.sampleRate = sampleRate;
            this.audioFormat = new AudioFormat(this.sampleRate, 16, 1, true, false);
            DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
            line = (SourceDataLine) AudioSystem.getLine(info);
            line.open(audioFormat);
            line.start();
            decoderThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        String b64Audio = b64AudioBuffer.poll();
                        if (b64Audio != null) {
                            byte[] rawAudio = Base64.getDecoder().decode(b64Audio);
                            RawAudioBuffer.add(rawAudio);
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            playerThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        byte[] rawAudio = RawAudioBuffer.poll();
                        if (rawAudio != null) {
                            try {
                                playChunk(rawAudio);
                            } catch (IOException e) {
                                throw new RuntimeException(e);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            decoderThread.start();
            playerThread.start();
        }

        // Play an audio chunk and block until playback is complete.
        private void playChunk(byte[] chunk) throws IOException, InterruptedException {
            if (chunk == null || chunk.length == 0) return;

            int bytesWritten = 0;
            while (bytesWritten < chunk.length) {
                bytesWritten += line.write(chunk, bytesWritten, chunk.length - bytesWritten);
            }
            int audioLength = chunk.length / (this.sampleRate*2/1000);
            // Wait for the audio in the buffer to finish playing.
            Thread.sleep(audioLength - 10);
        }

        public void write(String b64Audio) {
            b64AudioBuffer.add(b64Audio);
        }

        public void cancel() {
            b64AudioBuffer.clear();
            RawAudioBuffer.clear();
        }

        public void waitForComplete() throws InterruptedException {
            while (!b64AudioBuffer.isEmpty() || !RawAudioBuffer.isEmpty()) {
                Thread.sleep(100);
            }
            line.drain();
        }

        public void shutdown() throws InterruptedException {
            stopped.set(true);
            decoderThread.join();
            playerThread.join();
            if (line != null && line.isRunning()) {
                line.drain();
                line.close();
            }
        }
    }

    public static void main(String[] args) throws InterruptedException, LineUnavailableException, FileNotFoundException {
        QwenTtsRealtimeParam param = QwenTtsRealtimeParam.builder()
                .model("qwen3-tts-flash-realtime")
                // The following URL is for the Singapore region. If you use a model in the China (Beijing) region, replace the URL with wss://dashscope.aliyuncs.com/api-ws/v1/realtime.
                .url("wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime")
                // The API keys for the Singapore and China (Beijing) regions are different. To obtain an API key, see https://www.alibabacloud.com/help/en/model-studio/get-api-key.
                .apikey(System.getenv("DASHSCOPE_API_KEY"))
                .build();
        AtomicReference<CountDownLatch> completeLatch = new AtomicReference<>(new CountDownLatch(1));
        final AtomicReference<QwenTtsRealtime> qwenTtsRef = new AtomicReference<>(null);

        // Create a real-time audio player instance.
        RealtimePcmPlayer audioPlayer = new RealtimePcmPlayer(24000);

        QwenTtsRealtime qwenTtsRealtime = new QwenTtsRealtime(param, new QwenTtsRealtimeCallback() {
            @Override
            public void onOpen() {
                // Handle the event when the connection is established.
            }
            @Override
            public void onEvent(JsonObject message) {
                String type = message.get("type").getAsString();
                switch(type) {
                    case "session.created":
                        // Handle the event when the session is created.
                        break;
                    case "response.audio.delta":
                        String recvAudioB64 = message.get("delta").getAsString();
                        // Play the audio in real time.
                        audioPlayer.write(recvAudioB64);
                        break;
                    case "response.done":
                        // Handle the event when the response is complete.
                        break;
                    case "session.finished":
                        // Handle the event when the session is finished.
                        completeLatch.get().countDown();
                    default:
                        break;
                }
            }
            @Override
            public void onClose(int code, String reason) {
                // Handle the event when the connection is closed.
            }
        });
        qwenTtsRef.set(qwenTtsRealtime);
        try {
            qwenTtsRealtime.connect();
        } catch (NoApiKeyException e) {
            throw new RuntimeException(e);
        }
        QwenTtsRealtimeConfig config = QwenTtsRealtimeConfig.builder()
                .voice("Cherry")
                .responseFormat(QwenTtsRealtimeAudioFormat.PCM_24000HZ_MONO_16BIT)
                .mode("server_commit")
                .build();
        qwenTtsRealtime.updateSession(config);
        for (String text:textToSynthesize) {
            qwenTtsRealtime.appendText(text);
            Thread.sleep(100);
        }
        qwenTtsRealtime.finish();
        completeLatch.get().await();
        qwenTtsRealtime.close();

        // Wait for the audio to finish playing and then shut down the player.
        audioPlayer.waitForComplete();
        audioPlayer.shutdown();
        System.exit(0);
    }
}

// The Dashscope SDK version must be 2.21.16 or later.
import com.alibaba.dashscope.audio.qwen_tts_realtime.*;
import com.alibaba.dashscope.exception.NoApiKeyException;
import com.google.gson.JsonObject;
import javax.sound.sampled.LineUnavailableException;
import javax.sound.sampled.SourceDataLine;
import javax.sound.sampled.AudioFormat;
import javax.sound.sampled.DataLine;
import javax.sound.sampled.AudioSystem;
import java.io.File;
import java.io.FileNotFoundException;
import java.io.FileOutputStream;
import java.io.IOException;
import java.util.Base64;
import java.util.Queue;
import java.util.Scanner;
import java.util.concurrent.CountDownLatch;
import java.util.concurrent.atomic.AtomicReference;
import java.util.concurrent.ConcurrentLinkedQueue;
import java.util.concurrent.atomic.AtomicBoolean;

public class commit {
    // Real-time PCM audio player class
    public static class RealtimePcmPlayer {
        private int sampleRate;
        private SourceDataLine line;
        private AudioFormat audioFormat;
        private Thread decoderThread;
        private Thread playerThread;
        private AtomicBoolean stopped = new AtomicBoolean(false);
        private Queue<String> b64AudioBuffer = new ConcurrentLinkedQueue<>();
        private Queue<byte[]> RawAudioBuffer = new ConcurrentLinkedQueue<>();

        // The constructor initializes the audio format and audio line.
        public RealtimePcmPlayer(int sampleRate) throws LineUnavailableException {
            this.sampleRate = sampleRate;
            this.audioFormat = new AudioFormat(this.sampleRate, 16, 1, true, false);
            DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
            line = (SourceDataLine) AudioSystem.getLine(info);
            line.open(audioFormat);
            line.start();
            decoderThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        String b64Audio = b64AudioBuffer.poll();
                        if (b64Audio != null) {
                            byte[] rawAudio = Base64.getDecoder().decode(b64Audio);
                            RawAudioBuffer.add(rawAudio);
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            playerThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        byte[] rawAudio = RawAudioBuffer.poll();
                        if (rawAudio != null) {
                            try {
                                playChunk(rawAudio);
                            } catch (IOException e) {
                                throw new RuntimeException(e);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            decoderThread.start();
            playerThread.start();
        }

        // Play an audio chunk and block until playback is complete.
        private void playChunk(byte[] chunk) throws IOException, InterruptedException {
            if (chunk == null || chunk.length == 0) return;

            int bytesWritten = 0;
            while (bytesWritten < chunk.length) {
                bytesWritten += line.write(chunk, bytesWritten, chunk.length - bytesWritten);
            }
            int audioLength = chunk.length / (this.sampleRate*2/1000);
            // Wait for the audio in the buffer to finish playing.
            Thread.sleep(audioLength - 10);
        }

        public void write(String b64Audio) {
            b64AudioBuffer.add(b64Audio);
        }

        public void cancel() {
            b64AudioBuffer.clear();
            RawAudioBuffer.clear();
        }

        public void waitForComplete() throws InterruptedException {
            // Wait for all audio data in the buffers to finish playing.
            while (!b64AudioBuffer.isEmpty() || !RawAudioBuffer.isEmpty()) {
                Thread.sleep(100);
            }
            // Wait for the audio line to finish playing.
            line.drain();
        }

        public void shutdown() throws InterruptedException {
            stopped.set(true);
            decoderThread.join();
            playerThread.join();
            if (line != null && line.isRunning()) {
                line.drain();
                line.close();
            }
        }
    }

    public static void main(String[] args) throws InterruptedException, LineUnavailableException, FileNotFoundException {
        Scanner scanner = new Scanner(System.in);

        QwenTtsRealtimeParam param = QwenTtsRealtimeParam.builder()
                .model("qwen3-tts-flash-realtime")
                // The following URL is for the Singapore region. If you use a model in the China (Beijing) region, replace the URL with wss://dashscope.aliyuncs.com/api-ws/v1/realtime.
                .url("wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime")
                // The API keys for the Singapore and China (Beijing) regions are different. To obtain an API key, see https://www.alibabacloud.com/help/en/model-studio/get-api-key.
                .apikey(System.getenv("DASHSCOPE_API_KEY"))
                .build();

        AtomicReference<CountDownLatch> completeLatch = new AtomicReference<>(new CountDownLatch(1));

        // Create a real-time player instance.
        RealtimePcmPlayer audioPlayer = new RealtimePcmPlayer(24000);

        final AtomicReference<QwenTtsRealtime> qwenTtsRef = new AtomicReference<>(null);
        QwenTtsRealtime qwenTtsRealtime = new QwenTtsRealtime(param, new QwenTtsRealtimeCallback() {
//            File file = new File("result_24k.pcm");
//            FileOutputStream fos = new FileOutputStream(file);
            @Override
            public void onOpen() {
                System.out.println("connection opened");
                System.out.println("Enter text and press Enter to send. Enter 'quit' to exit the program.");
            }
            @Override
            public void onEvent(JsonObject message) {
                String type = message.get("type").getAsString();
                switch(type) {
                    case "session.created":
                        System.out.println("start session: " + message.get("session").getAsJsonObject().get("id").getAsString());
                        break;
                    case "response.audio.delta":
                        String recvAudioB64 = message.get("delta").getAsString();
                        byte[] rawAudio = Base64.getDecoder().decode(recvAudioB64);
                        //                            fos.write(rawAudio);
                        // Play the audio in real time.
                        audioPlayer.write(recvAudioB64);
                        break;
                    case "response.done":
                        System.out.println("response done");
                        // Wait for the audio to finish playing.
                        try {
                            audioPlayer.waitForComplete();
                        } catch (InterruptedException e) {
                            throw new RuntimeException(e);
                        }
                        // Prepare for the next input.
                        completeLatch.get().countDown();
                        break;
                    case "session.finished":
                        System.out.println("session finished");
                        if (qwenTtsRef.get() != null) {
                            System.out.println("[Metric] response: " + qwenTtsRef.get().getResponseId() +
                                    ", first audio delay: " + qwenTtsRef.get().getFirstAudioDelay() + " ms");
                        }
                        completeLatch.get().countDown();
                    default:
                        break;
                }
            }
            @Override
            public void onClose(int code, String reason) {
                System.out.println("connection closed code: " + code + ", reason: " + reason);
                try {
//                    fos.close();
                    // Wait for playback to complete and then shut down the player.
                    audioPlayer.waitForComplete();
                    audioPlayer.shutdown();
                } catch (InterruptedException e) {
                    throw new RuntimeException(e);
                }
            }
        });
        qwenTtsRef.set(qwenTtsRealtime);
        try {
            qwenTtsRealtime.connect();
        } catch (NoApiKeyException e) {
            throw new RuntimeException(e);
        }
        QwenTtsRealtimeConfig config = QwenTtsRealtimeConfig.builder()
                .voice("Cherry")
                .responseFormat(QwenTtsRealtimeAudioFormat.PCM_24000HZ_MONO_16BIT)
                .mode("commit")
                .build();
        qwenTtsRealtime.updateSession(config);

        // Loop to read user input.
        while (true) {
            System.out.print("Enter the text to synthesize: ");
            String text = scanner.nextLine();

            // If the user enters 'quit', exit the program.
            if ("quit".equalsIgnoreCase(text.trim())) {
                System.out.println("Closing the connection...");
                qwenTtsRealtime.finish();
                completeLatch.get().await();
                break;
            }

            // If the user input is empty, skip.
            if (text.trim().isEmpty()) {
                continue;
            }

            // Reinitialize the countdown latch.
            completeLatch.set(new CountDownLatch(1));

            // Send the text.
            qwenTtsRealtime.appendText(text);
            qwenTtsRealtime.commit();

            // Wait for the current synthesis to complete.
            completeLatch.get().await();
        }

        // Clean up resources.
        audioPlayer.waitForComplete();
        audioPlayer.shutdown();
        scanner.close();
        System.exit(0);
    }
}

WebSocket API

1. Prepare runtime environment

   Install pyaudio based on your operating system.

   macOS

   brew install portaudio && pip install pyaudio

   Debian/Ubuntu

   sudo apt-get install python3-pyaudio
   
   or
   
   pip install pyaudio

   CentOS

   sudo yum install -y portaudio portaudio-devel && pip install pyaudio

   Windows

   pip install pyaudio

   Then, install WebSocket dependencies using pip:

   pip install websocket-client==1.8.0 websockets

2. Create client

   Create a new Python file locally named tts_realtime_client.py and copy the following code into the file:

   tts_realtime_client.py

   # -- coding: utf-8 --
   
   import asyncio
   import websockets
   import json
   import base64
   import time
   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 TTS Realtime API.
   
       This class provides methods to connect to the TTS Realtime API, send text data, receive audio output, and manage WebSocket connections.
   
       Attributes:
           base_url (str):
               Base URL for the Realtime API.
           api_key (str):
               API Key for authentication.
           voice (str):
               Voice used by the server for speech synthesis.
           mode (SessionMode):
               Session mode, either server_commit or commit.
           audio_callback (Callable[[bytes], None]):
               Callback function to receive 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 status
           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 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,
               # Uncomment the lines below and replace model with qwen3-tts-instruct-flash-realtime in server_commit.py or commit.py to use instruction control
               # "instructions": "Speak quickly with a noticeably rising intonation, suitable for introducing fashion products.",
               # "optimize_instructions": true
               "language_type": self.language_type,
               "response_format": "pcm",
               "sample_rate": 24000
           })
   
   
       async def send_event(self, event) -> None:
           """Send event to server."""
           event['event_id'] = "event_" + str(int(time.time() * 1000))
           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 API."""
           event = {
               "type": "input_text_buffer.append",
               "text": text
           }
           await self.send_event(event)
   
   
       async def commit_text_buffer(self) -> None:
           """Submit text buffer to trigger processing."""
           event = {
               "type": "input_text_buffer.commit"
           }
           await self.send_event(event)
   
   
       async def clear_text_buffer(self) -> None:
           """Clear text buffer."""
           event = {
               "type": "input_text_buffer.clear"
           }
           await self.send_event(event)
   
   
       async def finish_session(self) -> None:
           """End session."""
           event = {
               "type": "session.finish"
           }
           await self.send_event(event)
   
   
       async def wait_for_response_done(self):
           """Wait for response.done event"""
           if self._response_done_future:
               await self._response_done_future
   
   
       async def handle_messages(self) -> None:
           """Handle messages from 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 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 complete
                       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 ended")
   
           except websockets.exceptions.ConnectionClosed:
               print("Connection closed")
           except Exception as e:
               print("Error handling messages: ", str(e))
   
   
       async def close(self) -> None:
           """Close WebSocket connection."""
           if self.ws:
               await self.ws.close()

3. Select speech synthesis mode

   The Realtime API supports two modes:

   • Server commit mode

     The client sends text only. The server intelligently determines text segmentation and synthesis timing. Use this mode for low-latency scenarios without manual synthesis control, such as GPS navigation.

   • Commit mode

     Add text to a buffer first, then trigger the server to synthesize the specified text. Use this mode for scenarios requiring fine-grained control over pauses and sentence breaks, such as news broadcasting.

   Server commit mode

   Create another Python file named server_commit.py in the same directory as tts_realtime_client.py, and copy the following code into the file:

   server_commit.py

   import os
   import asyncio
   import logging
   import wave
   from tts_realtime_client import TTSRealtimeClient, SessionMode
   import pyaudio
   
   # QwenTTS service configuration
   # Replace model with qwen3-tts-instruct-flash-realtime and uncomment instructions in tts_realtime_client.py to use instruction control
   # Singapore region URL. Replace with wss://dashscope.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime for Beijing region models
   URL = "wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime"
   # API Keys differ between Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
   # Replace with your Model Studio API Key if environment variable is not configured: 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 settings
   _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: play and cache in real time"""
       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 to 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 server"""
       text_fragments = [
           "Alibaba Cloud's Model Studio is an all-in-one platform for model development and application building.",
           "Both developers and business personnel can deeply participate in designing and building model applications.", 
           "You can develop a model application in just 5 minutes through simple UI operations,",
           "or train a custom model within hours, allowing you to focus more 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 server to complete internal processing before ending session
       await asyncio.sleep(1.0)
       await client.finish_session()
   
   async def _run_demo():
       """Run 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()
   
       # Execute 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() 

   Run server_commit.py to listen to real-time audio generated by the Realtime API.

   Commit mode

   Create another Python file named commit.py in the same directory as tts_realtime_client.py, and copy the following code into the file:

   commit.py

   import os
   import asyncio
   import logging
   import wave
   from tts_realtime_client import TTSRealtimeClient, SessionMode
   import pyaudio
   
   # QwenTTS service configuration
   # Replace model with qwen3-tts-instruct-flash-realtime and uncomment instructions in tts_realtime_client.py to use instruction control
   # Singapore region URL. Replace with wss://dashscope.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime for Beijing region models
   URL = "wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime?model=qwen3-tts-flash-realtime"
   # API Keys differ between Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
   # Replace with your Model Studio API Key if environment variable is not configured: 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: play and cache in real time"""
       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 to 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 commit event and end current session"""
       print("Enter text (press Enter directly to send commit event and end current session, press Ctrl+C or Ctrl+D to exit entire program):")
       
       while True:
           try:
               user_text = input("> ")
               if not user_text:  # User entered empty input
                   # Empty input signifies end of conversation: submit buffer -> end session -> break loop
                   logging.info("Empty input, sending commit event and ending current session")
                   await client.commit_text_buffer()
                   # Wait briefly for server to process commit to prevent losing audio from premature session end
                   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 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,  # Change to COMMIT mode
           audio_callback=_audio_callback
       )
   
       # Establish connection
       await client.connect()
   
       # Execute 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() 

   Run commit.py to input multiple texts for synthesis. Press Enter without entering text to listen to the audio returned by the Realtime API through your speakers.

Python

# coding=utf-8
# Installation instructions for pyaudio:
# APPLE Mac OS X
#   brew install portaudio
#   pip install pyaudio
# Debian/Ubuntu
#   sudo apt-get install python-pyaudio python3-pyaudio
#   or
#   pip install pyaudio
# CentOS
#   sudo yum install -y portaudio portaudio-devel && pip install pyaudio
# Microsoft Windows
#   python -m pip install pyaudio

import pyaudio
import os
import requests
import base64
import pathlib
import threading
import time
import dashscope  # DashScope Python SDK version must be at least 1.23.9
from dashscope.audio.qwen_tts_realtime import QwenTtsRealtime, QwenTtsRealtimeCallback, AudioFormat

# ======= Constants =======
DEFAULT_TARGET_MODEL = "qwen3-tts-vc-realtime-2026-01-15"  # Use the same model for voice cloning and speech synthesis
DEFAULT_PREFERRED_NAME = "guanyu"
DEFAULT_AUDIO_MIME_TYPE = "audio/mpeg"
VOICE_FILE_PATH = "voice.mp3"  # Relative path to local audio file for voice cloning

TEXT_TO_SYNTHESIZE = [
    'Right? I really love this kind of supermarket,',
    'especially during Chinese New Year',
    'when I go shopping',
    'I feel',
    'super super happy!',
    'I want to buy so many things!'
]

def create_voice(file_path: str,
                 target_model: str = DEFAULT_TARGET_MODEL,
                 preferred_name: str = DEFAULT_PREFERRED_NAME,
                 audio_mime_type: str = DEFAULT_AUDIO_MIME_TYPE) -> str:
    """
    Create voice and return voice parameter
    """
    # API Keys differ between Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
    # Replace with your Model Studio API Key if environment variable is not configured: api_key = "sk-xxx"
    api_key = os.getenv("DASHSCOPE_API_KEY")

    file_path_obj = pathlib.Path(file_path)
    if not file_path_obj.exists():
        raise FileNotFoundError(f"Audio file not found: {file_path}")

    base64_str = base64.b64encode(file_path_obj.read_bytes()).decode()
    data_uri = f"data:{audio_mime_type};base64,{base64_str}"

    # Singapore region URL. Replace with https://dashscope.aliyuncs.com/api/v1/services/audio/tts/customization for Beijing region models
    url = "https://dashscope-intl.aliyuncs.com/api/v1/services/audio/tts/customization"
    payload = {
        "model": "qwen-voice-enrollment", # Do not modify this value
        "input": {
            "action": "create",
            "target_model": target_model,
            "preferred_name": preferred_name,
            "audio": {"data": data_uri}
        }
    }
    headers = {
        "Authorization": f"Bearer {api_key}",
        "Content-Type": "application/json"
    }

    resp = requests.post(url, json=payload, headers=headers)
    if resp.status_code != 200:
        raise RuntimeError(f"Voice creation failed: {resp.status_code}, {resp.text}")

    try:
        return resp.json()["output"]["voice"]
    except (KeyError, ValueError) as e:
        raise RuntimeError(f"Failed to parse voice response: {e}")

def init_dashscope_api_key():
    """
    Initialize DashScope SDK API key
    """
    # API Keys differ between Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
    # Replace with your Model Studio API Key if environment variable is not configured: dashscope.api_key = "sk-xxx"
    dashscope.api_key = os.getenv("DASHSCOPE_API_KEY")

# ======= Callback class =======
class MyCallback(QwenTtsRealtimeCallback):
    """
    Custom TTS streaming callback
    """
    def __init__(self):
        self.complete_event = threading.Event()
        self._player = pyaudio.PyAudio()
        self._stream = self._player.open(
            format=pyaudio.paInt16, channels=1, rate=24000, output=True
        )

    def on_open(self) -> None:
        print('[TTS] Connection established')

    def on_close(self, close_status_code, close_msg) -> None:
        self._stream.stop_stream()
        self._stream.close()
        self._player.terminate()
        print(f'[TTS] Connection closed code={close_status_code}, msg={close_msg}')

    def on_event(self, response: dict) -> None:
        try:
            event_type = response.get('type', '')
            if event_type == 'session.created':
                print(f'[TTS] Session started: {response["session"]["id"]}')
            elif event_type == 'response.audio.delta':
                audio_data = base64.b64decode(response['delta'])
                self._stream.write(audio_data)
            elif event_type == 'response.done':
                print(f'[TTS] Response completed, Response ID: {qwen_tts_realtime.get_last_response_id()}')
            elif event_type == 'session.finished':
                print('[TTS] Session ended')
                self.complete_event.set()
        except Exception as e:
            print(f'[Error] Error handling callback event: {e}')

    def wait_for_finished(self):
        self.complete_event.wait()

# ======= Main execution logic =======
if __name__ == '__main__':
    init_dashscope_api_key()
    print('[System] Initializing Qwen TTS Realtime ...')

    callback = MyCallback()
    qwen_tts_realtime = QwenTtsRealtime(
        model=DEFAULT_TARGET_MODEL,
        callback=callback,
        # Singapore region URL. Replace with wss://dashscope.aliyuncs.com/api-ws/v1/realtime for Beijing region models
        url='wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime'
    )
    qwen_tts_realtime.connect()
    
    qwen_tts_realtime.update_session(
        voice=create_voice(VOICE_FILE_PATH), # Replace voice parameter with cloned custom voice
        response_format=AudioFormat.PCM_24000HZ_MONO_16BIT,
        mode='server_commit'
    )

    for text_chunk in TEXT_TO_SYNTHESIZE:
        print(f'[Sending text]: {text_chunk}')
        qwen_tts_realtime.append_text(text_chunk)
        time.sleep(0.1)

    qwen_tts_realtime.finish()
    callback.wait_for_finished()

    print(f'[Metric] session_id={qwen_tts_realtime.get_session_id()}, '
          f'first_audio_delay={qwen_tts_realtime.get_first_audio_delay()}s')

Java

<!-- https://mvnrepository.com/artifact/com.google.code.gson/gson -->
<dependency>
    <groupId>com.google.code.gson</groupId>
    <artifactId>gson</artifactId>
    <version>2.13.1</version>
</dependency>

// https://mvnrepository.com/artifact/com.google.code.gson/gson
implementation("com.google.code.gson:gson:2.13.1")

import com.alibaba.dashscope.audio.qwen_tts_realtime.*;
import com.alibaba.dashscope.exception.NoApiKeyException;
import com.google.gson.Gson;
import com.google.gson.JsonObject;

import javax.sound.sampled.*;
import java.io.*;
import java.net.HttpURLConnection;
import java.net.URL;
import java.nio.file.*;
import java.nio.charset.StandardCharsets;
import java.util.Base64;
import java.util.Queue;
import java.util.concurrent.CountDownLatch;
import java.util.concurrent.atomic.AtomicReference;
import java.util.concurrent.ConcurrentLinkedQueue;
import java.util.concurrent.atomic.AtomicBoolean;

public class Main {
    // ===== Constants =====
    // Use the same model for voice cloning and speech synthesis
    private static final String TARGET_MODEL = "qwen3-tts-vc-realtime-2026-01-15";
    private static final String PREFERRED_NAME = "guanyu";
    // Relative path to local audio file for voice cloning
    private static final String AUDIO_FILE = "voice.mp3";
    private static final String AUDIO_MIME_TYPE = "audio/mpeg";
    private static String[] textToSynthesize = {
            "Right? I really love this kind of supermarket",
            "especially during Chinese New Year",
            "when I go shopping",
            "I feel",
            "super super happy!",
            "I want to buy so many things!"
    };

    // Generate data URI
    public static String toDataUrl(String filePath) throws IOException {
        byte[] bytes = Files.readAllBytes(Paths.get(filePath));
        String encoded = Base64.getEncoder().encodeToString(bytes);
        return "data:" + AUDIO_MIME_TYPE + ";base64," + encoded;
    }

    // Call API to create voice
    public static String createVoice() throws Exception {
        // API Keys differ between Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
        // Replace with your Model Studio API Key if environment variable is not configured: String apiKey = "sk-xxx"
        String apiKey = System.getenv("DASHSCOPE_API_KEY");

        String jsonPayload =
                "{"
                        + "\"model\": \"qwen-voice-enrollment\"," // Do not modify this value
                        + "\"input\": {"
                        +     "\"action\": \"create\","
                        +     "\"target_model\": \"" + TARGET_MODEL + "\","
                        +     "\"preferred_name\": \"" + PREFERRED_NAME + "\","
                        +     "\"audio\": {"
                        +         "\"data\": \"" + toDataUrl(AUDIO_FILE) + "\""
                        +     "}"
                        + "}"
                        + "}";

        HttpURLConnection con = (HttpURLConnection) new URL("https://dashscope.aliyuncs.com/api/v1/services/audio/tts/customization").openConnection();
        con.setRequestMethod("POST");
        con.setRequestProperty("Authorization", "Bearer " + apiKey);
        con.setRequestProperty("Content-Type", "application/json");
        con.setDoOutput(true);

        try (OutputStream os = con.getOutputStream()) {
            os.write(jsonPayload.getBytes(StandardCharsets.UTF_8));
        }

        int status = con.getResponseCode();
        System.out.println("HTTP status code: " + status);

        try (BufferedReader br = new BufferedReader(
                new InputStreamReader(status >= 200 && status < 300 ? con.getInputStream() : con.getErrorStream(),
                        StandardCharsets.UTF_8))) {
            StringBuilder response = new StringBuilder();
            String line;
            while ((line = br.readLine()) != null) {
                response.append(line);
            }
            System.out.println("Response content: " + response);

            if (status == 200) {
                JsonObject jsonObj = new Gson().fromJson(response.toString(), JsonObject.class);
                return jsonObj.getAsJsonObject("output").get("voice").getAsString();
            }
            throw new IOException("Voice creation failed: " + status + " - " + response);
        }
    }

    // Real-time PCM audio player class
    public static class RealtimePcmPlayer {
        private int sampleRate;
        private SourceDataLine line;
        private AudioFormat audioFormat;
        private Thread decoderThread;
        private Thread playerThread;
        private AtomicBoolean stopped = new AtomicBoolean(false);
        private Queue<String> b64AudioBuffer = new ConcurrentLinkedQueue<>();
        private Queue<byte[]> RawAudioBuffer = new ConcurrentLinkedQueue<>();

        // Constructor to initialize audio format and audio line
        public RealtimePcmPlayer(int sampleRate) throws LineUnavailableException {
            this.sampleRate = sampleRate;
            this.audioFormat = new AudioFormat(this.sampleRate, 16, 1, true, false);
            DataLine.Info info = new DataLine.Info(SourceDataLine.class, audioFormat);
            line = (SourceDataLine) AudioSystem.getLine(info);
            line.open(audioFormat);
            line.start();
            decoderThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        String b64Audio = b64AudioBuffer.poll();
                        if (b64Audio != null) {
                            byte[] rawAudio = Base64.getDecoder().decode(b64Audio);
                            RawAudioBuffer.add(rawAudio);
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            playerThread = new Thread(new Runnable() {
                @Override
                public void run() {
                    while (!stopped.get()) {
                        byte[] rawAudio = RawAudioBuffer.poll();
                        if (rawAudio != null) {
                            try {
                                playChunk(rawAudio);
                            } catch (IOException e) {
                                throw new RuntimeException(e);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        } else {
                            try {
                                Thread.sleep(100);
                            } catch (InterruptedException e) {
                                throw new RuntimeException(e);
                            }
                        }
                    }
                }
            });
            decoderThread.start();
            playerThread.start();
        }

        // Play an audio chunk and block until playback completes
        private void playChunk(byte[] chunk) throws IOException, InterruptedException {
            if (chunk == null || chunk.length == 0) return;

            int bytesWritten = 0;
            while (bytesWritten < chunk.length) {
                bytesWritten += line.write(chunk, bytesWritten, chunk.length - bytesWritten);
            }
            int audioLength = chunk.length / (this.sampleRate*2/1000);
            // Wait for audio in buffer to finish playing
            Thread.sleep(audioLength - 10);
        }

        public void write(String b64Audio) {
            b64AudioBuffer.add(b64Audio);
        }

        public void cancel() {
            b64AudioBuffer.clear();
            RawAudioBuffer.clear();
        }

        public void waitForComplete() throws InterruptedException {
            while (!b64AudioBuffer.isEmpty() || !RawAudioBuffer.isEmpty()) {
                Thread.sleep(100);
            }
            line.drain();
        }

        public void shutdown() throws InterruptedException {
            stopped.set(true);
            decoderThread.join();
            playerThread.join();
            if (line != null && line.isRunning()) {
                line.drain();
                line.close();
            }
        }
    }

    public static void main(String[] args) throws Exception {
        QwenTtsRealtimeParam param = QwenTtsRealtimeParam.builder()
                .model(TARGET_MODEL)
                // Singapore region URL. Replace with wss://dashscope.aliyuncs.com/api-ws/v1/realtime for Beijing region models
                .url("wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime")
                // API Keys differ between Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
                // Replace with your Model Studio API Key if environment variable is not configured: .apikey("sk-xxx")
                .apikey(System.getenv("DASHSCOPE_API_KEY"))
                .build();
        AtomicReference<CountDownLatch> completeLatch = new AtomicReference<>(new CountDownLatch(1));
        final AtomicReference<QwenTtsRealtime> qwenTtsRef = new AtomicReference<>(null);

        // Create real-time audio player instance
        RealtimePcmPlayer audioPlayer = new RealtimePcmPlayer(24000);

        QwenTtsRealtime qwenTtsRealtime = new QwenTtsRealtime(param, new QwenTtsRealtimeCallback() {
            @Override
            public void onOpen() {
                // Handle connection established
            }
            @Override
            public void onEvent(JsonObject message) {
                String type = message.get("type").getAsString();
                switch(type) {
                    case "session.created":
                        // Handle session created
                        break;
                    case "response.audio.delta":
                        String recvAudioB64 = message.get("delta").getAsString();
                        // Play audio in real time
                        audioPlayer.write(recvAudioB64);
                        break;
                    case "response.done":
                        // Handle response completed
                        break;
                    case "session.finished":
                        // Handle session finished
                        completeLatch.get().countDown();
                    default:
                        break;
                }
            }
            @Override
            public void onClose(int code, String reason) {
                // Handle connection closed
            }
        });
        qwenTtsRef.set(qwenTtsRealtime);
        try {
            qwenTtsRealtime.connect();
        } catch (NoApiKeyException e) {
            throw new RuntimeException(e);
        }
        QwenTtsRealtimeConfig config = QwenTtsRealtimeConfig.builder()
                .voice(createVoice()) // Replace voice parameter with cloned custom voice
                .responseFormat(QwenTtsRealtimeAudioFormat.PCM_24000HZ_MONO_16BIT)
                .mode("server_commit")
                .build();
        qwenTtsRealtime.updateSession(config);
        for (String text:textToSynthesize) {
            qwenTtsRealtime.appendText(text);
            Thread.sleep(100);
        }
        qwenTtsRealtime.finish();
        completeLatch.get().await();

        // Wait for audio playback to complete and shut down player
        audioPlayer.waitForComplete();
        audioPlayer.shutdown();
        System.exit(0);
    }
}