Stream text-to-speech conversion with low first-packet latency. Real-time speech synthesis supports streaming input and output, voice cloning, voice design, and fine-grained audio controls for voice assistants, audiobooks, and intelligent customer service.
Overview
Convert text to speech in real time with low latency.
- Streaming input and output with low first-packet latency
- Adjustable speech rate, pitch, volume, and bitrate for fine-grained audio control
- Compatible with mainstream audio formats (PCM, WAV, MP3, Opus) with up to 48 kHz sample rate output
- Supports Instruction control, which lets you control speech expressiveness through natural language instructions
- Supports Voice cloning and Voice Design for custom voice creation
- Supports Emotion and rich language tags, which lets you embed emotion or sound effect tags within text
For batch scenarios such as audiobooks and courseware voiceover, use Non-real-time speech synthesis. For model selection guidance, see Speech synthesis.
Prerequisites
- Configure an API key and set it as an environment variable.
- If you call the API through the DashScope SDK, install the latest SDK.
- To use the AOQ protocol with CosyVoice models, download and integrate the AOQ client SDK. For details, see SDK overview.
Quick start
The following examples demonstrate speech synthesis for each model. For more examples and parameter details, see the API reference.
Qwen-Audio-TTS
The following example synthesizes speech using a system voice.
To use the Instruction control feature, set instructions through the instruction parameter.
# coding=utf-8
import os
import dashscope
from dashscope.audio.tts_v2 import *
# The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
# If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: dashscope.api_key = "sk-xxx"
dashscope.api_key = os.environ.get('DASHSCOPE_API_KEY')
# The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
dashscope.base_websocket_api_url='wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference'
# Model
# qwen-audio-3.0-tts-flash/qwen-audio-3.0-tts-plus: Use voices such as longanhuan_v3.6.
# Each voice supports different languages. To synthesize non-Chinese languages such as Japanese or Korean, select a voice that supports the target language. See the voice list for details.
model = "qwen-audio-3.0-tts-flash"
# Voice
voice = "longanhuan_v3.6"
# Instantiate SpeechSynthesizer and pass request parameters such as model and voice in the constructor
synthesizer = SpeechSynthesizer(model=model, voice=voice)
# Send the text to be synthesized and get the binary audio
audio = synthesizer.call("How is the weather today?")
# The first text submission requires establishing a WebSocket connection, so the first-packet latency includes connection setup time
print('[Metric] requestId: {}, first-packet latency: {} ms'.format(
synthesizer.get_last_request_id(),
synthesizer.get_first_package_delay()))
# Save the audio to a local file
with open('output.mp3', 'wb') as f:
f.write(audio)
import com.alibaba.dashscope.audio.ttsv2.SpeechSynthesisParam;
import com.alibaba.dashscope.audio.ttsv2.SpeechSynthesizer;
import com.alibaba.dashscope.utils.Constants;
import java.io.File;
import java.io.FileOutputStream;
import java.io.IOException;
import java.nio.ByteBuffer;
public class Main {
// Model
// qwen-audio-3.0-tts-flash/qwen-audio-3.0-tts-plus: Use voices such as longanhuan_v3.6.
// Each voice supports different languages. To synthesize non-Chinese languages such as Japanese or Korean, select a voice that supports the target language. See the voice list for details.
private static String model = "qwen-audio-3.0-tts-flash";
// Voice
private static String voice = "longanhuan_v3.6";
public static void streamAudioDataToSpeaker() {
// Request parameters
SpeechSynthesisParam param =
SpeechSynthesisParam.builder()
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: .apiKey("sk-xxx")
.apiKey(System.getenv("DASHSCOPE_API_KEY"))
.model(model) // Model
.voice(voice) // Voice
.build();
// Synchronous mode: disable callback (second parameter is null)
SpeechSynthesizer synthesizer = new SpeechSynthesizer(param, null);
ByteBuffer audio = null;
try {
// Block until audio is returned
audio = synthesizer.call("How is the weather today?");
} catch (Exception e) {
throw new RuntimeException(e);
} finally {
// Close the WebSocket connection when the task is done
synthesizer.getDuplexApi().close(1000, "bye");
}
if (audio != null) {
// Save the audio data to the local file "output.mp3"
File file = new File("output.mp3");
// The first text submission requires establishing a WebSocket connection, so the first-packet latency includes connection setup time
// Note: getFirstPackageDelay() requires dashscope-sdk-java 2.18.0 or later
System.out.println(
"[Metric] requestId: "
+ synthesizer.getLastRequestId()
+ ", first-packet latency (ms): "
+ synthesizer.getFirstPackageDelay());
try (FileOutputStream fos = new FileOutputStream(file)) {
fos.write(audio.array());
} catch (IOException e) {
throw new RuntimeException(e);
}
}
}
public static void main(String[] args) {
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
Constants.baseWebsocketApiUrl = "wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference";
streamAudioDataToSpeaker();
System.exit(0);
}
}
CosyVoice
In addition to WebSocket, this model also supports the AOQ protocol. For client-side integration that prioritizes stable latency, resilience on weak networks, and built-in full-duplex noise suppression and echo cancellation, AOQ is recommended. For a protocol comparison, see Realtime API overview.
Importantcosyvoice-v3.5-plus and cosyvoice-v3.5-flash are available only in the Beijing region and support only voice design and voice cloning scenarios (no system voices). Before use, create a custom voice through Voice cloning or Voice Design, then set voice to the voice ID and model to the corresponding model name in your code.
The following example synthesizes speech using a system voice (see CosyVoice Voice list).
To use the Instruction control feature, set instructions through the instruction parameter.
# coding=utf-8
import os
import dashscope
from dashscope.audio.tts_v2 import *
# The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
# If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: dashscope.api_key = "sk-xxx"
dashscope.api_key = os.environ.get('DASHSCOPE_API_KEY')
# The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
dashscope.base_websocket_api_url='wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference'
# Model
# Different model versions require their corresponding voices:
# cosyvoice-v3-flash/cosyvoice-v3-plus: Use voices such as longanyang.
# cosyvoice-v2: Use voices such as longxiaochun_v2.
# Each voice supports different languages. To synthesize non-Chinese languages such as Japanese or Korean, select a voice that supports the target language. See the Qwen-Audio-TTS/CosyVoice voice list for details.
model = "cosyvoice-v3-flash"
# Voice
voice = "longanyang"
# Instantiate SpeechSynthesizer and pass request parameters such as model and voice in the constructor
synthesizer = SpeechSynthesizer(model=model, voice=voice)
# Send the text to be synthesized and get the binary audio
audio = synthesizer.call("How is the weather today?")
# The first text submission requires establishing a WebSocket connection, so the first-packet latency includes connection setup time
print('[Metric] requestId: {}, first-packet latency: {} ms'.format(
synthesizer.get_last_request_id(),
synthesizer.get_first_package_delay()))
# Save the audio to a local file
with open('output.mp3', 'wb') as f:
f.write(audio)
import com.alibaba.dashscope.audio.ttsv2.SpeechSynthesisParam;
import com.alibaba.dashscope.audio.ttsv2.SpeechSynthesizer;
import com.alibaba.dashscope.utils.Constants;
import java.io.File;
import java.io.FileOutputStream;
import java.io.IOException;
import java.nio.ByteBuffer;
public class Main {
// Model
// Different model versions require their corresponding voices:
// cosyvoice-v3-flash/cosyvoice-v3-plus: Use voices such as longanyang.
// cosyvoice-v2: Use voices such as longxiaochun_v2.
// Each voice supports different languages. To synthesize non-Chinese languages such as Japanese or Korean, select a voice that supports the target language. See the Qwen-Audio-TTS/CosyVoice voice list for details.
private static String model = "cosyvoice-v3-flash";
// Voice
private static String voice = "longanyang";
public static void streamAudioDataToSpeaker() {
// Request parameters
SpeechSynthesisParam param =
SpeechSynthesisParam.builder()
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: .apiKey("sk-xxx")
.apiKey(System.getenv("DASHSCOPE_API_KEY"))
.model(model) // Model
.voice(voice) // Voice
.build();
// Synchronous mode: disable callback (second parameter is null)
SpeechSynthesizer synthesizer = new SpeechSynthesizer(param, null);
ByteBuffer audio = null;
try {
// Block until audio is returned
audio = synthesizer.call("How is the weather today?");
} catch (Exception e) {
throw new RuntimeException(e);
} finally {
// Close the WebSocket connection when the task is done
synthesizer.getDuplexApi().close(1000, "bye");
}
if (audio != null) {
// Save the audio data to the local file "output.mp3"
File file = new File("output.mp3");
// The first text submission requires establishing a WebSocket connection, so the first-packet latency includes connection setup time
// Note: getFirstPackageDelay() requires dashscope-sdk-java 2.18.0 or later
System.out.println(
"[Metric] requestId: "
+ synthesizer.getLastRequestId()
+ ", first-packet latency (ms): "
+ synthesizer.getFirstPackageDelay());
try (FileOutputStream fos = new FileOutputStream(file)) {
fos.write(audio.array());
} catch (IOException e) {
throw new RuntimeException(e);
}
}
}
public static void main(String[] args) {
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
Constants.baseWebsocketApiUrl = "wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference";
streamAudioDataToSpeaker();
System.exit(0);
}
}
Qwen-TTS
The following example synthesizes speech using a system voice (see Supported voices).
To use the Instruction control feature, replace model with qwen3-tts-instruct-flash-realtime and set instructions through the instructions parameter.
Python
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/en/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,
# Singapore region
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/en/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,
# Singapore region
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(),
))
Java
Server commit mode
appendText()
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.*;
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 really love this kind of supermarket.",
"Especially during the Chinese New Year.",
"Going to the supermarket.",
"It just makes me feel.",
"Super, super happy!",
"I want to buy so many things!"
};
public static QwenTtsRealtimeAudioFormat ttsFormat = QwenTtsRealtimeAudioFormat.PCM_24000HZ_MONO_16BIT;
// Real-time PCM audio player
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<>();
private ByteArrayOutputStream totalAudioStream = new ByteArrayOutputStream();
// Initialize 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);
// Write audio data to totalAudioStream.
try {
totalAudioStream.write(rawAudio);
} catch (IOException e) {
throw new RuntimeException(e);
}
} 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 the buffered audio 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, IOException {
stopped.set(true);
decoderThread.join();
playerThread.join();
// Save the complete audio file.
File file = new File("TotalAudio_"+ttsFormat.getSampleRate()+"."+ttsFormat.getFormat());
try (FileOutputStream fos = new FileOutputStream(file)) {
fos.write(totalAudioStream.toByteArray());
}
if (line != null && line.isRunning()) {
line.drain();
line.close();
}
}
}
public static void main(String[] args) throws InterruptedException, LineUnavailableException, IOException {
QwenTtsRealtimeParam param = QwenTtsRealtimeParam.builder()
// To use instruction control, replace the model with qwen3-tts-instruct-flash-realtime.
.model("qwen3-tts-flash-realtime")
// China (Beijing) region
.url("wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime")
// API keys differ between Singapore and China (Beijing). 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 connection establishment.
}
@Override
public void onEvent(JsonObject message) {
String type = message.get("type").getAsString();
switch(type) {
case "session.created":
// Handle session creation.
if (message.has("session")) {
String eventId = message.get("event_id").getAsString();
String sessionId = message.get("session").getAsJsonObject().get("id").getAsString();
System.out.println("[onEvent] session.created, session_id: "
+ sessionId + ", event_id: " + eventId);
}
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 completion.
break;
case "session.finished":
// Handle session termination.
completeLatch.get().countDown();
default:
break;
}
}
@Override
public void onClose(int code, String reason) {
// Handle connection closure.
}
});
qwenTtsRef.set(qwenTtsRealtime);
try {
qwenTtsRealtime.connect();
} catch (NoApiKeyException e) {
throw new RuntimeException(e);
}
QwenTtsRealtimeConfig config = QwenTtsRealtimeConfig.builder()
.voice("Cherry")
.responseFormat(ttsFormat)
.mode("server_commit")
// To use instruction control, uncomment the following lines and replace the model with qwen3-tts-instruct-flash-realtime.
// .instructions("")
// .optimizeInstructions(true)
.build();
qwenTtsRealtime.updateSession(config);
for (String text:textToSynthesize) {
qwenTtsRealtime.appendText(text);
Thread.sleep(100);
}
qwenTtsRealtime.finish();
completeLatch.get().await();
qwenTtsRealtime.close();
// Wait for audio playback to complete, then shut down the player.
audioPlayer.waitForComplete();
audioPlayer.shutdown();
System.exit(0);
}
}
Commit mode
commit()
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.*;
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 Main {
public static QwenTtsRealtimeAudioFormat ttsFormat = QwenTtsRealtimeAudioFormat.PCM_24000HZ_MONO_16BIT;
// Real-time PCM audio player
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<>();
private ByteArrayOutputStream totalAudioStream = new ByteArrayOutputStream();
// Initialize 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);
// Write audio data to totalAudioStream.
try {
totalAudioStream.write(rawAudio);
} catch (IOException e) {
throw new RuntimeException(e);
}
} 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 the buffered audio 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 buffered audio data to finish playing.
while (!b64AudioBuffer.isEmpty() || !RawAudioBuffer.isEmpty()) {
Thread.sleep(100);
}
// Wait for the audio line to drain.
line.drain();
}
public void shutdown() throws InterruptedException {
stopped.set(true);
decoderThread.join();
playerThread.join();
// Save the complete audio file.
File file = new File("TotalAudio_"+ttsFormat.getSampleRate()+"."+ttsFormat.getFormat());
try (FileOutputStream fos = new FileOutputStream(file)) {
fos.write(totalAudioStream.toByteArray());
} catch (FileNotFoundException e) {
throw new RuntimeException(e);
} catch (IOException e) {
throw new RuntimeException(e);
}
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()
// To use instruction control, replace the model with qwen3-tts-instruct-flash-realtime.
.model("qwen3-tts-flash-realtime")
// China (Beijing) region
.url("wss://dashscope-intl.aliyuncs.com/api-ws/v1/realtime")
// API keys differ between Singapore and China (Beijing). 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() {
@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);
// Play audio in real time.
audioPlayer.write(recvAudioB64);
break;
case "response.done":
System.out.println("response done");
// Wait for audio playback to complete.
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 {
// Wait for playback to complete, 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(ttsFormat)
.mode("commit")
// To use instruction control, uncomment the following lines and replace the model with qwen3-tts-instruct-flash-realtime.
// .instructions("")
// .optimizeInstructions(true)
.build();
qwenTtsRealtime.updateSession(config);
// Read user input in a loop.
while (true) {
System.out.print("Enter the text to synthesize: ");
String text = scanner.nextLine();
// Exit when the user enters 'quit'.
if ("quit".equalsIgnoreCase(text.trim())) {
System.out.println("Closing the connection...");
qwenTtsRealtime.finish();
completeLatch.get().await();
break;
}
// Skip empty input.
if (text.trim().isEmpty()) {
continue;
}
// Re-initialize 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);
}
}
Session configuration
Qwen-TTS interaction modes
Qwen-TTS Realtime API provides two interaction modes:
- server_commit mode: The server handles text segmentation and synthesis timing automatically. Suited for continuous synthesis of large text blocks. The client appends text without managing segmentation or submission.
- commit mode: The client explicitly submits the text buffer to trigger synthesis. Suited for scenarios that require precise control over synthesis timing, such as per-turn synthesis in conversational AI.
Switch interaction modes:
- WebSocket: Set the
session.updateevent'smodefield.
{
"type": "session.update",
"session": {
"mode": "server_commit"
}
}
- Python SDK: Set the
modeparameter in theupdate_sessionmethod.
qwen_tts_realtime.update_session(
voice='Cherry',
response_format=AudioFormat.PCM_24000HZ_MONO_16BIT,
mode='server_commit'
)
- Java SDK: Set the
modeparameter throughQwenTtsRealtimeConfig.builder().
QwenTtsRealtimeConfig config = QwenTtsRealtimeConfig.builder()
.voice("Cherry")
.responseFormat(ttsFormat)
.mode("server_commit")
.build();
qwenTtsRealtime.updateSession(config);
For complete SDK code examples, see Python SDK and Java SDK. For WebSocket event lifecycle and connection reuse details, see WebSocket API reference.
Advanced features
Instruction control
Instruction control uses natural language descriptions to adjust speech tone, speed, emotion, and timbre characteristics without configuring complex audio parameters.
Instruction specifications by model:
Qwen-Audio-TTS
Supported models: qwen-audio-3.0-tts-plus, qwen-audio-3.0-tts-flash
System voices and voice cloning voices: accept any instruction.
CosyVoice
Supported models: cosyvoice-v3.5-plus, cosyvoice-v3.5-flash, cosyvoice-v3-plus, cosyvoice-v3-flash
Instruction format requirements vary by model:
-
cosyvoice-v3.5-plus, cosyvoice-v3.5-flash:
- Voice cloning/design voices: accept any instruction.
- System voices: v3.5 doesn't support system voices.
-
cosyvoice-v3-plus:
- Voice cloning/design voices: don't support instruction control.
- System voices: instructions must use a fixed format and content. See CosyVoice Voice list.
-
cosyvoice-v3-flash:
- Voice cloning/design voices: accept any instruction.
- System voices: instructions must use a fixed format and content. See CosyVoice Voice list.
Usage: Specify instruction content through the instruction parameter.
Supported languages for instruction text:
-
cosyvoice-v3.5-plus, cosyvoice-v3.5-flash:
- Voice cloning/design voices: Chinese, English, French, German, Japanese, Korean, Russian, Portuguese, Thai, Indonesian, and Vietnamese.
- System voices: v3.5 doesn't support system voices.
-
cosyvoice-v3-plus:
- Voice cloning/design voices: Chinese, English, French, German, Japanese, Korean, and Russian.
- System voices: instructions must use a fixed format and content. See CosyVoice Voice list.
-
cosyvoice-v3-flash:
- Voice cloning/design voices: Chinese, English, French, German, Japanese, Korean, and Russian.
- System voices: Chinese only.
Instruction text length limit: 100 characters maximum. Chinese characters (including simplified/traditional Chinese, Japanese kanji, and Korean hanja) count as 2 characters each. All other characters (such as punctuation, letters, digits, Japanese kana, and Korean hangul) count as 1 character each.
Qwen-TTS
Supported models: Qwen3-TTS-Instruct-Flash-Realtime series models only.
Usage: Specify instruction content through the instructions parameter.
Supported languages for instruction text: Chinese and English only.
Instruction text length limit: 1,600 tokens maximum.
Use cases:
- Audiobook and radio drama voiceover
- Advertising and promotional video voiceover
- Game character and animation voiceover
- Emotionally expressive voice assistants
- Documentary and news broadcast narration
Writing effective voice descriptions:
-
Core principles:
- Be specific, not vague: Use words that depict vocal qualities, such as "deep", "crisp", or "slightly fast pace". Avoid subjective or vague terms like "nice" or "normal".
- Be multi-dimensional, not one-dimensional: A good description typically covers multiple dimensions (such as gender, age, and emotion). Writing only "female voice" is too broad to generate a distinctive timbre.
- Be objective, not subjective: Focus on the physical and perceptual characteristics of the voice. For example, use "higher pitch with an energetic tone" instead of "my favorite voice".
- Be original, not imitative: Describe the vocal qualities rather than requesting imitation of specific individuals (such as celebrities or actors). The model doesn't support imitation, and it may pose copyright risks.
- Be concise, not redundant: Make sure every word serves a purpose. Avoid repetitive synonyms or meaningless modifiers.
-
Description dimension reference:
Combine the following dimensions to describe a voice. The more dimensions you include, the more accurate the output.
Dimension
Example descriptions
Gender
Male, female, androgynous
Age
Child (5-12), teenager (13-18), young adult (19-35), middle-aged (36-55), elderly (55+)
Pitch
High, mid-range, low, slightly high, slightly low
Speed
Fast, moderate, slow, slightly fast, slightly slow
Emotion
Cheerful, calm, gentle, serious, lively, composed, soothing
Characteristics
Magnetic, crisp, husky, mellow, sweet, rich, powerful
Use case
News broadcast, ad voiceover, audiobook, animation character, voice assistant, documentary narration
-
Examples:
- Standard broadcast style: clear and precise articulation with perfect pronunciation
- A young, lively female voice with a faster pace and noticeable upward intonation, suited for fashion product introductions
- A calm middle-aged male, slow pace, deep and magnetic voice, suited for news reading or documentary narration
- A gentle, intellectual female, around 30 years old, with an even tone, suited for audiobook narration
- A cute child's voice, approximately an 8-year-old girl, speaking with a slightly childish quality, suited for animation character voiceover
Dialects
This section describes how to produce speech in Chinese dialects (such as Henan dialect, Sichuan dialect, and Cantonese). Configuration methods vary by model and voice type.
Dialect configuration by model:
Qwen-Audio-TTS
-
System voices: Select one of the following voice types:
- A system voice with built-in dialect support, which outputs the corresponding dialect without additional configuration.
- A voice that supports Instruction control and can be configured to output a specific dialect through instruction text.
-
Voice cloning voices: Configure through the Instruction control feature. For example, set the instruction text to
请用河南话表达.
Supported dialects: See the "Supported languages" column for each model in Qwen-Audio-TTS.
CosyVoice
-
System voices: Select one of the following voice types from CosyVoice Voice list:
- A system voice with built-in dialect support (such as
longshange_v3), which outputs the corresponding dialect without additional configuration. - A voice that supports Instruction control and can be configured to output a specific dialect (such as
longanhuan_v3), specified through instruction text.
- A system voice with built-in dialect support (such as
-
Voice cloning voices: Configure through the Instruction control feature. For example, set the instruction text to
请用河南话表达. -
Voice design voices: don't support dialects.
Supported dialects: See the "Supported languages" column for each model in CosyVoice.
Example: Use cosyvoice-v3-flash with the longanhuan_v3 voice, and set the instruction text to "请用河南话表达。" to output speech in Henan dialect.
# coding=utf-8
import os
import dashscope
from dashscope.audio.tts_v2 import *
# The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
# If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: dashscope.api_key = "sk-xxx"
dashscope.api_key = os.environ.get('DASHSCOPE_API_KEY')
# The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
dashscope.base_websocket_api_url='wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference'
# Model
# Different model versions require their corresponding voices:
# cosyvoice-v3-flash/cosyvoice-v3-plus: Use voices such as longanyang.
# cosyvoice-v2: Use voices such as longxiaochun_v2.
# Select the appropriate voice for your target language
model = "cosyvoice-v3-flash"
# Voice
voice = "longanhuan_v3"
# Instantiate SpeechSynthesizer and pass request parameters such as model and voice in the constructor
synthesizer = SpeechSynthesizer(model=model, voice=voice, instruction="请用河南话表达。")
# Send the text to be synthesized and get the binary audio
audio = synthesizer.call("叫你去买盐,你买回来一袋面,这不是弄啥嘞吗!")
# The first text submission requires establishing a WebSocket connection, so the first-packet latency includes connection setup time
print('[Metric] requestId: {}, first-packet latency: {} ms'.format(
synthesizer.get_last_request_id(),
synthesizer.get_first_package_delay()))
# Save the audio to a local file
with open('output.mp3', 'wb') as f:
f.write(audio)
Qwen-TTS
- System voices: Use a system voice with built-in dialect support. See the Qwen-TTS voice list in Supported voices.
- Voice cloning voices: don't support dialects.
- Voice design voices: don't support dialects.
Supported dialects: See the "Supported languages" column for each model in Qwen3-TTS.
Emotion and rich language tags
Qwen-Audio-TTS series models support embedding emotion and rich language tags directly in the text to synthesize (the text parameter). These tags control emotional expression or insert vocal effects (such as laughter and sighs) at specified positions, producing more expressive speech without configuring complex audio parameters.
ImportantSupported models: qwen-audio-3.0-tts-plus and qwen-audio-3.0-tts-flash only.
Limitation: Only unidirectional streaming mode is supported.
Control tags set the emotion or style of the speech. Place a tag in the text to affect all subsequent text until the next control tag appears or the sentence is automatically segmented due to length.
Tag | Description |
|---|---|
| Sad |
| Amazed |
| Deep, loud shouting |
| Trembling |
| Angry |
| Excited |
| Sarcastic |
| Curious |
| Dracula style (deep, eerie) |
| Bored |
| Tired |
| Scornful |
| Shouting |
| ASMR soft whisper |
| Panicked |
| Mischievous |
| Empathetic |
| Whisper |
| Reluctant |
| Crying |
| Serious |
| Very slow speech |
| Very fast speech |
Rich language tags insert a vocal effect at the current position in the text without affecting the emotional style of surrounding text.
Tag | Description |
|---|---|
| Gasp |
| Sigh |
| Throat clearing |
| Giggle |
| Laughter |
| Cough |
| Snort |
The following example shows how to combine control tags and rich language tags in the text parameter:
[excited]What a beautiful day today![laughing]Let's go out and have fun together!
In this text, [excited] is a control tag that applies an excited emotion to all subsequent text. [laughing] is a rich language tag that inserts a laugh at that position before continuing to synthesize the remaining text.
You can also switch between different emotions within the same text:
[serious]Please pay attention to the safety precautions.[excited]Alright, let's get started now!
Here, [serious] sets the first sentence to a serious tone, and [excited] switches to an excited tone starting from the second sentence.
Cancel task
If you need to interrupt the current synthesis turn during real-time speech synthesis, send a cancel command. After cancellation, the server immediately ends the current task and returns a completion event. You can start a new synthesis task on the same WebSocket connection without reconnecting.
Usage:
- Python SDK: 1.26.4 or later, call
SpeechSynthesizer.streaming_cancel(). - Java SDK: 2.22.26 or later, call
SpeechSynthesizer.streamingCancel(). - WebSocket raw protocol: Send a
finish-taskevent and setdirective=cancelininput.
ImportantModel limitations:
- China (Beijing): All Qwen-Audio-TTS models support this feature. CosyVoice models require v2 or later.
- Singapore: All Qwen-Audio-TTS models support this feature. CosyVoice models do not support this feature.
WebSocket raw protocol calls
The following examples demonstrate how to connect directly to the server through the WebSocket raw protocol, suited for scenarios without the DashScope SDK. These are minimal runnable implementations. For WebSocket protocol details, see the API reference for each model.
View WebSocket raw protocol call examples
Qwen-Audio-TTS/CosyVoice
Qwen-Audio-TTS and Qwen-Audio-TTS/CosyVoice use the same WebSocket protocol. The following examples use qwen-audio-3.0-tts-flash. To use Qwen-Audio-TTS/CosyVoice, replace the model parameter with a Qwen-Audio-TTS/CosyVoice model (such as cosyvoice-v3-flash) and the voice parameter with the desired voice.
Go
package main
import (
"encoding/json"
"fmt"
"net/http"
"os"
"strings"
"time"
"github.com/google/uuid"
"github.com/gorilla/websocket"
)
const (
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
wsURL = "wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference"
outputFile = "output.mp3"
)
func main() {
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: apiKey := "sk-xxx"
apiKey := os.Getenv("DASHSCOPE_API_KEY")
// Clear the output file
os.Remove(outputFile)
os.Create(outputFile)
// Connect to WebSocket
header := make(http.Header)
header.Add("X-DashScope-DataInspection", "enable")
header.Add("Authorization", fmt.Sprintf("bearer %s", apiKey))
conn, resp, err := websocket.DefaultDialer.Dial(wsURL, header)
if err != nil {
if resp != nil {
fmt.Printf("Connection failed, HTTP status code: %d\n", resp.StatusCode)
}
fmt.Println("Connection failed:", err)
return
}
defer conn.Close()
// Generate task ID
taskID := uuid.New().String()
fmt.Printf("Generated task ID: %s\n", taskID)
// Sending run-task event
runTaskCmd := map[string]interface{}{
"header": map[string]interface{}{
"action": "run-task",
"task_id": taskID,
"streaming": "duplex",
},
"payload": map[string]interface{}{
"task_group": "audio",
"task": "tts",
"function": "SpeechSynthesizer",
"model": "qwen-audio-3.0-tts-flash",
"parameters": map[string]interface{}{
"text_type": "PlainText",
"voice": "longanhuan_v3.6",
"format": "mp3",
"sample_rate": 22050,
"volume": 50,
"rate": 1,
"pitch": 1,
// If enable_ssml is set to true, only one continue-task event can be sent; otherwise the error "Text request limit violated, expected 1." will be returned
"enable_ssml": false,
},
"input": map[string]interface{}{},
},
}
runTaskJSON, _ := json.Marshal(runTaskCmd)
fmt.Printf("Sending run-task event: %s\n", string(runTaskJSON))
err = conn.WriteMessage(websocket.TextMessage, runTaskJSON)
if err != nil {
fmt.Println("Failed to send run-task:", err)
return
}
textSent := false
// Process messages
for {
messageType, message, err := conn.ReadMessage()
if err != nil {
fmt.Println("Failed to read message:", err)
break
}
// Process binary message
if messageType == websocket.BinaryMessage {
fmt.Printf("Received binary message, length: %d\n", len(message))
file, _ := os.OpenFile(outputFile, os.O_APPEND|os.O_WRONLY|os.O_CREATE, 0644)
file.Write(message)
file.Close()
continue
}
// Process text message
messageStr := string(message)
fmt.Printf("Received text message: %s\n", strings.ReplaceAll(messageStr, "\n", ""))
// Parse JSON to get event type
var msgMap map[string]interface{}
if json.Unmarshal(message, &msgMap) == nil {
if header, ok := msgMap["header"].(map[string]interface{}); ok {
if event, ok := header["event"].(string); ok {
fmt.Printf("Event type: %s\n", event)
switch event {
case "task-started":
fmt.Println("=== Received task-started event ===")
if !textSent {
// Sending continue-task event
texts := []string{"Before my bed, moonlight shines bright, I suspect it's frost upon the ground.", "I raise my eyes to gaze at the bright moon, then bow my head, thinking of home."}
for _, text := range texts {
continueTaskCmd := map[string]interface{}{
"header": map[string]interface{}{
"action": "continue-task",
"task_id": taskID,
"streaming": "duplex",
},
"payload": map[string]interface{}{
"input": map[string]interface{}{
"text": text,
},
},
}
continueTaskJSON, _ := json.Marshal(continueTaskCmd)
fmt.Printf("Sending continue-task event: %s\n", string(continueTaskJSON))
err = conn.WriteMessage(websocket.TextMessage, continueTaskJSON)
if err != nil {
fmt.Println("Failed to send continue-task:", err)
return
}
}
textSent = true
// Delay before sending finish-task
time.Sleep(500 * time.Millisecond)
// Sending finish-task event
finishTaskCmd := map[string]interface{}{
"header": map[string]interface{}{
"action": "finish-task",
"task_id": taskID,
"streaming": "duplex",
},
"payload": map[string]interface{}{
"input": map[string]interface{}{},
},
}
finishTaskJSON, _ := json.Marshal(finishTaskCmd)
fmt.Printf("Sending finish-task event: %s\n", string(finishTaskJSON))
err = conn.WriteMessage(websocket.TextMessage, finishTaskJSON)
if err != nil {
fmt.Println("Failed to send finish-task:", err)
return
}
}
case "task-finished":
fmt.Println("=== Task completed ===")
return
case "task-failed":
fmt.Println("=== Task failed ===")
if header["error_message"] != nil {
fmt.Printf("Error message: %s\n", header["error_message"])
}
return
case "result-generated":
fmt.Println("Received result-generated event")
}
}
}
}
}
}
C#
using System.Net.WebSockets;
using System.Text;
using System.Text.Json;
class Program {
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: private static readonly string ApiKey = "sk-xxx"
private static readonly string ApiKey = Environment.GetEnvironmentVariable("DASHSCOPE_API_KEY") ?? throw new InvalidOperationException("DASHSCOPE_API_KEY environment variable is not set.");
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
private const string WebSocketUrl = "wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference";
// Output file path
private const string OutputFilePath = "output.mp3";
// WebSocket client
private static ClientWebSocket _webSocket = new ClientWebSocket();
// Cancellation token source
private static CancellationTokenSource _cancellationTokenSource = new CancellationTokenSource();
// Task ID
private static string? _taskId;
// Whether the task has started
private static TaskCompletionSource<bool> _taskStartedTcs = new TaskCompletionSource<bool>();
static async Task Main(string[] args) {
try {
// Clear the output file
ClearOutputFile(OutputFilePath);
// Connect to the WebSocket service
await ConnectToWebSocketAsync(WebSocketUrl);
// Start the message receiving task
Task receiveTask = ReceiveMessagesAsync();
// Send the run-task event
_taskId = GenerateTaskId();
await SendRunTaskCommandAsync(_taskId);
// Wait for the task-started event
await _taskStartedTcs.Task;
// Send continue-task events
string[] texts = {
"Before my bed, moonlight shines bright,",
"I suspect it\'s frost upon the ground.",
"I raise my eyes to gaze at the bright moon,",
"then bow my head, thinking of home."
};
foreach (string text in texts) {
await SendContinueTaskCommandAsync(text);
}
// Send the finish-task event
await SendFinishTaskCommandAsync(_taskId);
// Wait for the receiving task to complete
await receiveTask;
Console.WriteLine("Task completed, connection closed.");
} catch (OperationCanceledException) {
Console.WriteLine("Task cancelled.");
} catch (Exception ex) {
Console.WriteLine($"Error occurred: {ex.Message}");
} finally {
_cancellationTokenSource.Cancel();
_webSocket.Dispose();
}
}
private static void ClearOutputFile(string filePath) {
if (File.Exists(filePath)) {
File.WriteAllText(filePath, string.Empty);
Console.WriteLine("Output file cleared.");
} else {
Console.WriteLine("Output file does not exist, no need to clear.");
}
}
private static async Task ConnectToWebSocketAsync(string url) {
var uri = new Uri(url);
if (_webSocket.State == WebSocketState.Connecting || _webSocket.State == WebSocketState.Open) {
return;
}
// Set the WebSocket connection headers
_webSocket.Options.SetRequestHeader("Authorization", $"bearer {ApiKey}");
_webSocket.Options.SetRequestHeader("X-DashScope-DataInspection", "enable");
try {
await _webSocket.ConnectAsync(uri, _cancellationTokenSource.Token);
Console.WriteLine("Successfully connected to the WebSocket service.");
} catch (OperationCanceledException) {
Console.WriteLine("WebSocket connection cancelled.");
} catch (Exception ex) {
Console.WriteLine($"WebSocket connection failed: {ex.Message}");
throw;
}
}
private static async Task SendRunTaskCommandAsync(string taskId) {
var command = CreateCommand("run-task", taskId, "duplex", new {
task_group = "audio",
task = "tts",
function = "SpeechSynthesizer",
model = "qwen-audio-3.0-tts-flash",
parameters = new
{
text_type = "PlainText",
voice = "longanhuan_v3.6",
format = "mp3",
sample_rate = 22050,
volume = 50,
rate = 1,
pitch = 1,
// If enable_ssml is set to true, only one continue-task event can be sent; otherwise the error "Text request limit violated, expected 1." will be returned
enable_ssml = false
},
input = new { }
});
await SendJsonMessageAsync(command);
Console.WriteLine("run-task event sent.");
}
private static async Task SendContinueTaskCommandAsync(string text) {
if (_taskId == null) {
throw new InvalidOperationException("Task ID not initialized.");
}
var command = CreateCommand("continue-task", _taskId, "duplex", new {
input = new {
text
}
});
await SendJsonMessageAsync(command);
Console.WriteLine("continue-task event sent.");
}
private static async Task SendFinishTaskCommandAsync(string taskId) {
var command = CreateCommand("finish-task", taskId, "duplex", new {
input = new { }
});
await SendJsonMessageAsync(command);
Console.WriteLine("finish-task event sent.");
}
private static async Task SendJsonMessageAsync(string message) {
var buffer = Encoding.UTF8.GetBytes(message);
try {
await _webSocket.SendAsync(new ArraySegment<byte>(buffer), WebSocketMessageType.Text, true, _cancellationTokenSource.Token);
} catch (OperationCanceledException) {
Console.WriteLine("Message sending cancelled.");
}
}
private static async Task ReceiveMessagesAsync() {
while (_webSocket.State == WebSocketState.Open) {
var response = await ReceiveMessageAsync();
if (response != null) {
var eventStr = response.RootElement.GetProperty("header").GetProperty("event").GetString();
switch (eventStr) {
case "task-started":
Console.WriteLine("Task started.");
_taskStartedTcs.TrySetResult(true);
break;
case "task-finished":
Console.WriteLine("Task completed.");
_cancellationTokenSource.Cancel();
break;
case "task-failed":
Console.WriteLine("Task failed: " + response.RootElement.GetProperty("header").GetProperty("error_message").GetString());
_cancellationTokenSource.Cancel();
break;
default:
// result-generated can be handled here
break;
}
}
}
}
private static async Task<JsonDocument?> ReceiveMessageAsync() {
var buffer = new byte[1024 * 4];
var segment = new ArraySegment<byte>(buffer);
try {
WebSocketReceiveResult result = await _webSocket.ReceiveAsync(segment, _cancellationTokenSource.Token);
if (result.MessageType == WebSocketMessageType.Close) {
await _webSocket.CloseAsync(WebSocketCloseStatus.NormalClosure, "Closing", _cancellationTokenSource.Token);
return null;
}
if (result.MessageType == WebSocketMessageType.Binary) {
// Process binary data
Console.WriteLine("Received binary data...");
// Save the binary data to file
using (var fileStream = new FileStream(OutputFilePath, FileMode.Append)) {
fileStream.Write(buffer, 0, result.Count);
}
return null;
}
string message = Encoding.UTF8.GetString(buffer, 0, result.Count);
return JsonDocument.Parse(message);
} catch (OperationCanceledException) {
Console.WriteLine("Message receiving cancelled.");
return null;
}
}
private static string GenerateTaskId() {
return Guid.NewGuid().ToString("N").Substring(0, 32);
}
private static string CreateCommand(string action, string taskId, string streaming, object payload) {
var command = new {
header = new {
action,
task_id = taskId,
streaming
},
payload
};
return JsonSerializer.Serialize(command);
}
}
PHP
The sample code directory structure:
my-php-project/
├── composer.json
├── vendor/
└── index.php
composer.json contents (adjust dependency versions as needed):
{
"require": {
"react/event-loop": "^1.3",
"react/socket": "^1.11",
"react/stream": "^1.2",
"react/http": "^1.1",
"ratchet/pawl": "^0.4"
},
"autoload": {
"psr-4": {
"App\\": "src/"
}
}
}
index.php contents:
<?php
require __DIR__ . '/vendor/autoload.php';
use Ratchet\Client\Connector;
use React\EventLoop\Loop;
use React\Socket\Connector as SocketConnector;
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: $api_key = "sk-xxx"
$api_key = getenv("DASHSCOPE_API_KEY");
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
$websocket_url = 'wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference'; // WebSocket server URL
$output_file = 'output.mp3'; // Output file path
$loop = Loop::get();
if (file_exists($output_file)) {
// Clear file content
file_put_contents($output_file, '');
}
// Create a custom connector
$socketConnector = new SocketConnector($loop, [
'tcp' => [
'bindto' => '0.0.0.0:0',
],
'tls' => [
'verify_peer' => false,
'verify_peer_name' => false,
],
]);
$connector = new Connector($loop, $socketConnector);
$headers = [
'Authorization' => 'bearer ' . $api_key,
'X-DashScope-DataInspection' => 'enable'
];
$connector($websocket_url, [], $headers)->then(function ($conn) use ($loop, $output_file) {
echo "Connected to the WebSocket server\n";
// Generate task ID
$taskId = generateTaskId();
// Send the run-task event
sendRunTaskMessage($conn, $taskId);
// Define the function to send continue-task events
$sendContinueTask = function() use ($conn, $loop, $taskId) {
// Text to be sent
$texts = ["Before my bed, moonlight shines bright,", "I suspect it\'s frost upon the ground.", "I raise my eyes to gaze at the bright moon,", "then bow my head, thinking of home."];
$continueTaskCount = 0;
foreach ($texts as $text) {
$continueTaskMessage = json_encode([
"header" => [
"action" => "continue-task",
"task_id" => $taskId,
"streaming" => "duplex"
],
"payload" => [
"input" => [
"text" => $text
]
]
]);
echo "Sending continue-task event: " . $continueTaskMessage . "\n";
$conn->send($continueTaskMessage);
$continueTaskCount++;
}
echo "Number of continue-task events sent: " . $continueTaskCount . "\n";
// Send the finish-task event
sendFinishTaskMessage($conn, $taskId);
};
// Flag for whether the task-started event was received
$taskStarted = false;
// Listen for messages
$conn->on('message', function($msg) use ($conn, $sendContinueTask, $loop, &$taskStarted, $taskId, $output_file) {
if ($msg->isBinary()) {
// Write binary data to local file
file_put_contents($output_file, $msg->getPayload(), FILE_APPEND);
} else {
// Process non-binary message
$response = json_decode($msg, true);
if (isset($response['header']['event'])) {
handleEvent($conn, $response, $sendContinueTask, $loop, $taskId, $taskStarted);
} else {
echo "Unknown message format\n";
}
}
});
// Listen for connection close
$conn->on('close', function($code = null, $reason = null) {
echo "Connection closed\n";
if ($code !== null) {
echo "Close code: " . $code . "\n";
}
if ($reason !== null) {
echo "Close reason: " . $reason . "\n";
}
});
}, function ($e) {
echo "Unable to connect: {$e->getMessage()}\n";
});
$loop->run();
/**
* Generate task ID
* @return string
*/
function generateTaskId(): string {
return bin2hex(random_bytes(16));
}
/**
* Send the run-task event
* @param $conn
* @param $taskId
*/
function sendRunTaskMessage($conn, $taskId) {
$runTaskMessage = json_encode([
"header" => [
"action" => "run-task",
"task_id" => $taskId,
"streaming" => "duplex"
],
"payload" => [
"task_group" => "audio",
"task" => "tts",
"function" => "SpeechSynthesizer",
"model" => "qwen-audio-3.0-tts-flash",
"parameters" => [
"text_type" => "PlainText",
"voice" => "longanhuan_v3.6",
"format" => "mp3",
"sample_rate" => 22050,
"volume" => 50,
"rate" => 1,
"pitch" => 1,
// If enable_ssml is set to true, only one continue-task event can be sent; otherwise the error "Text request limit violated, expected 1." will be returned
"enable_ssml" => false
],
"input" => (object) []
]
]);
echo "Sending run-task event: " . $runTaskMessage . "\n";
$conn->send($runTaskMessage);
echo "run-task event sent\n";
}
/**
* Read audio file
* @param string $filePath
* @return bool|string
*/
function readAudioFile(string $filePath) {
$voiceData = file_get_contents($filePath);
if ($voiceData === false) {
echo "Unable to read audio file\n";
}
return $voiceData;
}
/**
* Split audio data
* @param string $data
* @param int $chunkSize
* @return array
*/
function splitAudioData(string $data, int $chunkSize): array {
return str_split($data, $chunkSize);
}
/**
* Send the finish-task event
* @param $conn
* @param $taskId
*/
function sendFinishTaskMessage($conn, $taskId) {
$finishTaskMessage = json_encode([
"header" => [
"action" => "finish-task",
"task_id" => $taskId,
"streaming" => "duplex"
],
"payload" => [
"input" => (object) []
]
]);
echo "Sending finish-task event: " . $finishTaskMessage . "\n";
$conn->send($finishTaskMessage);
echo "finish-task event sent\n";
}
/**
* Handle events
* @param $conn
* @param $response
* @param $sendContinueTask
* @param $loop
* @param $taskId
* @param $taskStarted
*/
function handleEvent($conn, $response, $sendContinueTask, $loop, $taskId, &$taskStarted) {
switch ($response['header']['event']) {
case 'task-started':
echo "Task started, sending continue-task events...\n";
$taskStarted = true;
// Send continue-task events
$sendContinueTask();
break;
case 'result-generated':
// Received result-generated event
break;
case 'task-finished':
echo "Task completed\n";
$conn->close();
break;
case 'task-failed':
echo "Task failed\n";
echo "Error code: " . $response['header']['error_code'] . "\n";
echo "Error message: " . $response['header']['error_message'] . "\n";
$conn->close();
break;
case 'error':
echo "Error: " . $response['payload']['message'] . "\n";
break;
default:
echo "Unknown event: " . $response['header']['event'] . "\n";
break;
}
// If the task is completed, close the connection
if ($response['header']['event'] == 'task-finished') {
// Wait 1 second to ensure all data has been transmitted
$loop->addTimer(1, function() use ($conn) {
$conn->close();
echo "Client closing connection\n";
});
}
// If the task-started event has not been received, close the connection
if (!$taskStarted && in_array($response['header']['event'], ['task-failed', 'error'])) {
$conn->close();
}
}
Node.js
Install the required dependencies:
npm install ws
npm install uuid
Sample code:
const WebSocket = require('ws');
const fs = require('fs');
const uuid = require('uuid').v4;
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: const apiKey = "sk-xxx"
const apiKey = process.env.DASHSCOPE_API_KEY;
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
const url = 'wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference';
// Output file path
const outputFilePath = 'output.mp3';
// Clear output file
fs.writeFileSync(outputFilePath, '');
// Create WebSocket client
const ws = new WebSocket(url, {
headers: {
Authorization: `bearer ${apiKey}`,
'X-DashScope-DataInspection': 'enable'
}
});
let taskStarted = false;
let taskId = uuid();
ws.on('open', () => {
console.log('Connected to the WebSocket server');
// Send the run-task event
const runTaskMessage = JSON.stringify({
header: {
action: 'run-task',
task_id: taskId,
streaming: 'duplex'
},
payload: {
task_group: 'audio',
task: 'tts',
function: 'SpeechSynthesizer',
model: 'qwen-audio-3.0-tts-flash',
parameters: {
text_type: 'PlainText',
voice: 'longanhuan_v3.6', // Voice
format: 'mp3', // Audio format
sample_rate: 22050, // Sample rate
volume: 50, // Volume
rate: 1, // Speech rate
pitch: 1, // Pitch
enable_ssml: false // Whether to enable SSML. If enable_ssml is set to true, only one continue-task event can be sent; otherwise the error "Text request limit violated, expected 1." will be returned
},
input: {}
}
});
ws.send(runTaskMessage);
console.log('run-task message sent');
});
const fileStream = fs.createWriteStream(outputFilePath, { flags: 'a' });
ws.on('message', (data, isBinary) => {
if (isBinary) {
// Write binary data to file
fileStream.write(data);
} else {
const message = JSON.parse(data);
switch (message.header.event) {
case 'task-started':
taskStarted = true;
console.log('Task started');
// Send continue-task events
sendContinueTasks(ws);
break;
case 'task-finished':
console.log('Task completed');
ws.close();
fileStream.end(() => {
console.log('File stream closed');
});
break;
case 'task-failed':
console.error('Task failed: ', message.header.error_message);
ws.close();
fileStream.end(() => {
console.log('File stream closed');
});
break;
default:
// result-generated can be handled here
break;
}
}
});
function sendContinueTasks(ws) {
const texts = [
'Before my bed, moonlight shines bright,',
'I suspect it is frost upon the ground.',
'I raise my eyes to gaze at the bright moon,',
'then bow my head, thinking of home.'
];
texts.forEach((text, index) => {
setTimeout(() => {
if (taskStarted) {
const continueTaskMessage = JSON.stringify({
header: {
action: 'continue-task',
task_id: taskId,
streaming: 'duplex'
},
payload: {
input: {
text: text
}
}
});
ws.send(continueTaskMessage);
console.log(`continue-task sent, text: ${text}`);
}
}, index * 1000); // Send one every second
});
// Send the finish-task event
setTimeout(() => {
if (taskStarted) {
const finishTaskMessage = JSON.stringify({
header: {
action: 'finish-task',
task_id: taskId,
streaming: 'duplex'
},
payload: {
input: {}
}
});
ws.send(finishTaskMessage);
console.log('finish-task sent');
}
}, texts.length * 1000 + 1000); // Send 1 second after all continue-task events are sent
}
ws.on('close', () => {
console.log('Disconnected from the WebSocket server');
});
Java
We recommend using the Java DashScope SDK for development. See Java SDK.
The following is a Java WebSocket direct connection example. Import these dependencies before running:
Java-WebSocketjackson-databind
Use Maven or Gradle to manage dependencies:
<dependencies>
<!-- WebSocket Client -->
<dependency>
<groupId>org.java-websocket</groupId>
<artifactId>Java-WebSocket</artifactId>
<version>1.5.3</version>
</dependency>
<!-- JSON Processing -->
<dependency>
<groupId>com.fasterxml.jackson.core</groupId>
<artifactId>jackson-databind</artifactId>
<version>2.13.0</version>
</dependency>
</dependencies>
// Other code omitted
dependencies {
// WebSocket Client
implementation 'org.java-websocket:Java-WebSocket:1.5.3'
// JSON Processing
implementation 'com.fasterxml.jackson.core:jackson-databind:2.13.0'
}
// Other code omitted
Java code:
import com.fasterxml.jackson.databind.ObjectMapper;
import org.java_websocket.client.WebSocketClient;
import org.java_websocket.handshake.ServerHandshake;
import java.io.FileOutputStream;
import java.io.IOException;
import java.net.URI;
import java.nio.ByteBuffer;
import java.util.*;
public class TTSWebSocketClient extends WebSocketClient {
private final String taskId = UUID.randomUUID().toString();
private final String outputFile = "output_" + System.currentTimeMillis() + ".mp3";
private boolean taskFinished = false;
public TTSWebSocketClient(URI serverUri, Map<String, String> headers) {
super(serverUri, headers);
}
@Override
public void onOpen(ServerHandshake serverHandshake) {
System.out.println("Connection established");
// Send the run-task event
// If enable_ssml is set to true, only one continue-task event can be sent; otherwise the error "Text request limit violated, expected 1." will be returned
String runTaskCommand = "{ \"header\": { \"action\": \"run-task\", \"task_id\": \"" + taskId + "\", \"streaming\": \"duplex\" }, \"payload\": { \"task_group\": \"audio\", \"task\": \"tts\", \"function\": \"SpeechSynthesizer\", \"model\": \"qwen-audio-3.0-tts-flash\", \"parameters\": { \"text_type\": \"PlainText\", \"voice\": \"longanhuan_v3.6\", \"format\": \"mp3\", \"sample_rate\": 22050, \"volume\": 50, \"rate\": 1, \"pitch\": 1, \"enable_ssml\": false }, \"input\": {} }}";
send(runTaskCommand);
}
@Override
public void onMessage(String message) {
System.out.println("Received message from server: " + message);
try {
// Parse JSON message
Map<String, Object> messageMap = new ObjectMapper().readValue(message, Map.class);
if (messageMap.containsKey("header")) {
Map<String, Object> header = (Map<String, Object>) messageMap.get("header");
if (header.containsKey("event")) {
String event = (String) header.get("event");
if ("task-started".equals(event)) {
System.out.println("Received task-started event from server");
List<String> texts = Arrays.asList(
"Before my bed, moonlight shines bright,I suspect it\'s frost upon the ground.",
"I raise my eyes to gaze at the bright moon,then bow my head, thinking of home."
);
for (String text : texts) {
// Send the continue-task event
sendContinueTask(text);
}
// Send the finish-task event
sendFinishTask();
} else if ("task-finished".equals(event)) {
System.out.println("Received task-finished event from server");
taskFinished = true;
closeConnection();
} else if ("task-failed".equals(event)) {
System.out.println("Task failed: " + message);
closeConnection();
}
}
}
} catch (Exception e) {
System.err.println("Exception occurred: " + e.getMessage());
}
}
@Override
public void onMessage(ByteBuffer message) {
System.out.println("Received binary audio data of size: " + message.remaining());
try (FileOutputStream fos = new FileOutputStream(outputFile, true)) {
byte[] buffer = new byte[message.remaining()];
message.get(buffer);
fos.write(buffer);
System.out.println("Audio data written to local file " + outputFile);
} catch (IOException e) {
System.err.println("Failed to write audio data to local file: " + e.getMessage());
}
}
@Override
public void onClose(int code, String reason, boolean remote) {
System.out.println("Connection closed: " + reason + " (" + code + ")");
}
@Override
public void onError(Exception ex) {
System.err.println("Error: " + ex.getMessage());
ex.printStackTrace();
}
private void sendContinueTask(String text) {
String command = "{ \"header\": { \"action\": \"continue-task\", \"task_id\": \"" + taskId + "\", \"streaming\": \"duplex\" }, \"payload\": { \"input\": { \"text\": \"" + text + "\" } }}";
send(command);
}
private void sendFinishTask() {
String command = "{ \"header\": { \"action\": \"finish-task\", \"task_id\": \"" + taskId + "\", \"streaming\": \"duplex\" }, \"payload\": { \"input\": {} }}";
send(command);
}
private void closeConnection() {
if (!isClosed()) {
close();
}
}
public static void main(String[] args) {
try {
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: String apiKey = "sk-xxx"
String apiKey = System.getenv("DASHSCOPE_API_KEY");
if (apiKey == null || apiKey.isEmpty()) {
System.err.println("Please set the DASHSCOPE_API_KEY environment variable");
return;
}
Map<String, String> headers = new HashMap<>();
headers.put("Authorization", "bearer " + apiKey);
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
TTSWebSocketClient client = new TTSWebSocketClient(new URI("wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference"), headers);
client.connect();
while (!client.isClosed() && !client.taskFinished) {
Thread.sleep(1000);
}
} catch (Exception e) {
System.err.println("Failed to connect to WebSocket service: " + e.getMessage());
e.printStackTrace();
}
}
}
Python
We recommend using the Python DashScope SDK for development. See Python SDK.
The following is a Python WebSocket direct connection example. Install these dependencies before running:
pip uninstall websocket-client
pip uninstall websocket
pip install websocket-client
ImportantDon't name the Python file "websocket.py", otherwise an error occurs (AttributeError: module 'websocket' has no attribute 'WebSocketApp'. Did you mean: 'WebSocket'?).
import websocket
import json
import uuid
import os
import time
class TTSClient:
def __init__(self, api_key, uri):
"""
Initialize TTSClient instance
Parameters:
api_key (str): API Key for authentication
uri (str): WebSocket service URL
"""
self.api_key = api_key # Replace with your API Key
self.uri = uri # Replace with your WebSocket URL
self.task_id = str(uuid.uuid4()) # Generate unique task ID
self.output_file = f"output_{int(time.time())}.mp3" # Output audio file path
self.ws = None # WebSocketApp instance
self.task_started = False # Whether task-started has been received
self.task_finished = False # Whether task-finished / task-failed has been received
def on_open(self, ws):
"""
Callback when WebSocket connection is established
Sends run-task event to start speech synthesis task
"""
print("WebSocket connected")
# Construct run-task event
run_task_cmd = {
"header": {
"action": "run-task",
"task_id": self.task_id,
"streaming": "duplex"
},
"payload": {
"task_group": "audio",
"task": "tts",
"function": "SpeechSynthesizer",
"model": "qwen-audio-3.0-tts-flash",
"parameters": {
"text_type": "PlainText",
"voice": "longanhuan_v3.6",
"format": "mp3",
"sample_rate": 22050,
"volume": 50,
"rate": 1,
"pitch": 1,
# If enable_ssml is set to true, only one continue-task event can be sent; otherwise an error will be returned
"enable_ssml": False
},
"input": {}
}
}
# Send run-task event
ws.send(json.dumps(run_task_cmd))
print("Sent run-task event")
def on_message(self, ws, message):
"""
Callback when a message is received
Handles text and binary messages differently
"""
if isinstance(message, str):
# Handle JSON text messages
try:
msg_json = json.loads(message)
print(f"Received JSON message: {msg_json}")
if "header" in msg_json:
header = msg_json["header"]
if "event" in header:
event = header["event"]
if event == "task-started":
print("Task started")
self.task_started = True
# Send continue-task events
texts = [
"Before my bed, moonlight shines bright,I suspect it\'s frost upon the ground.",
"I raise my eyes to gaze at the bright moon,then bow my head, thinking of home."
]
for text in texts:
self.send_continue_task(text)
# Send finish-task after all continue-task events are sent
self.send_finish_task()
elif event == "task-finished":
print("Task completed")
self.task_finished = True
self.close(ws)
elif event == "task-failed":
error_msg = msg_json.get("error_message", "Unknown error")
print(f"Task failed: {error_msg}")
self.task_finished = True
self.close(ws)
except json.JSONDecodeError as e:
print(f"JSON parsing failed: {e}")
else:
# Handle binary messages (audio data)
print(f"Received binary message, size: {len(message)} bytes")
with open(self.output_file, "ab") as f:
f.write(message)
print(f"Audio data written to local file {self.output_file}")
def on_error(self, ws, error):
"""Callback when an error occurs"""
print(f"WebSocket error: {error}")
def on_close(self, ws, close_status_code, close_msg):
"""Callback when connection is closed"""
print(f"WebSocket closed: {close_msg} ({close_status_code})")
def send_continue_task(self, text):
"""Send continue-task event with text content to be synthesized"""
cmd = {
"header": {
"action": "continue-task",
"task_id": self.task_id,
"streaming": "duplex"
},
"payload": {
"input": {
"text": text
}
}
}
self.ws.send(json.dumps(cmd))
print(f"Sent continue-task event, text content: {text}")
def send_finish_task(self):
"""Send finish-task event to end the speech synthesis task"""
cmd = {
"header": {
"action": "finish-task",
"task_id": self.task_id,
"streaming": "duplex"
},
"payload": {
"input": {}
}
}
self.ws.send(json.dumps(cmd))
print("Sent finish-task event")
def close(self, ws):
"""Actively close the connection"""
if ws and ws.sock and ws.sock.connected:
ws.close()
print("Connection closed actively")
def run(self):
"""Start the WebSocket client"""
# Set request headers (authentication)
header = {
"Authorization": f"bearer {self.api_key}",
"X-DashScope-DataInspection": "enable"
}
# Create WebSocketApp instance
self.ws = websocket.WebSocketApp(
self.uri,
header=header,
on_open=self.on_open,
on_message=self.on_message,
on_error=self.on_error,
on_close=self.on_close
)
print("Listening for WebSocket messages...")
self.ws.run_forever() # Start persistent connection listener
# Example usage
if __name__ == "__main__":
# The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
# If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: API_KEY = "sk-xxx"
API_KEY = os.environ.get("DASHSCOPE_API_KEY")
# The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
SERVER_URI = "wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference" # Replace with your WebSocket URL
client = TTSClient(API_KEY, SERVER_URI)
client.run()
Qwen-TTS
-
Create the client
Python
Create a Python file named
tts_realtime_client.pyand copy the following code into the file:# -- 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 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, # To use the instruction control feature, uncomment the lines below and replace the model with qwen3-tts-instruct-flash-realtime in server_commit.py or commit.py # "instructions": "Speak quickly with a noticeable 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 an event to the 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 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()Java
Create a Java file named
TTSRealtimeClient.javaand copy the following code into the file: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); // To use the instruction control feature, uncomment the lines below and replace the model with qwen3-tts-instruct-flash-realtime // session.addProperty("instructions", "Speak quickly with a noticeable rising intonation, suitable for introducing fashion products."); // session.addProperty("optimize_instructions", true); 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_" + System.currentTimeMillis(); 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(); } } } -
Select a speech synthesis mode
Realtime API supports two modes:
-
server_commit mode
The server handles text segmentation and synthesis timing automatically. The client only sends text. Suited for low-latency scenarios (such as GPS navigation).
-
commit mode
The client adds text to a buffer and explicitly triggers synthesis. Suited for scenarios that require precise sentence segmentation control (such as news broadcast).
server_commit mode
Python
In the same directory as
tts_realtime_client.py, create another Python file namedserver_commit.pyand copy the following code into the file:import os import asyncio import logging import wave from tts_realtime_client import TTSRealtimeClient, SessionMode import pyaudio # QwenTTS service configuration # To use the instruction control feature, replace the model with qwen3-tts-instruct-flash-realtime and uncomment instructions in tts_realtime_client.py # 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/zh/model-studio/get-api-key # If you have not configured the environment variable, replace the next line with your Chinese 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()Run
server_commit.pyto hear the audio generated by Realtime API in real time.Java
In the same directory as
TTSRealtimeClient.java, create another Java file namedServerCommit.javaand copy the following code into the file: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/zh/model-studio/get-api-key // If you have not configured the environment variable, replace the next line with your Chinese 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 // To use the instruction control feature, replace the model with qwen3-tts-instruct-flash-realtime and uncomment the instructions in TTSRealtimeClient.java TTSRealtimeClient client = new TTSRealtimeClient( URL, API_KEY, "Cherry", TTSRealtimeClient.SessionMode.SERVER_COMMIT, audioData -> { playbackQueue.add(audioData); audioChunks.add(audioData); System.out.println("Received audio data: " + audioData.length + " bytes"); } ); client.connect(); // Send text fragments String[] textFragments = { "Alibaba Cloud's large language model platform, Model Studio, is an all-in-one platform for developing and building large language model applications.", "Both developers and business users can deeply participate in the design and development of large language model applications.", "You can develop a large language model application in five minutes using a simple interface,", "or train a dedicated model in a few hours, allowing you to focus more energy on application innovation." }; System.out.println("Starting to send text..."); for (String text : textFragments) { System.out.println("Sending fragment: " + text); client.appendText(text); Thread.sleep(100); } Thread.sleep(1000); client.finishSession(); // Wait for the response to complete client.waitForResponseDone(); client.waitForSessionFinished(); client.close(); // Wait for playback to complete playing.set(false); playerThread.join(); audioLine.drain(); audioLine.close(); // Save the audio file saveWav("output.wav"); System.out.println("Done"); } private static void saveWav(String filename) throws IOException { if (audioChunks.isEmpty()) { System.out.println("No audio data to save"); return; } ByteArrayOutputStream bos = new ByteArrayOutputStream(); for (byte[] chunk : audioChunks) { bos.write(chunk); } byte[] allAudio = bos.toByteArray(); AudioFormat format = new AudioFormat(SAMPLE_RATE, 16, 1, true, false); AudioInputStream ais = new AudioInputStream( new ByteArrayInputStream(allAudio), format, allAudio.length / 2); new File("outputs").mkdirs(); AudioSystem.write(ais, AudioFileFormat.Type.WAVE, new File("outputs/" + filename)); System.out.println("Audio saved to: outputs/" + filename); } }Compile and run
ServerCommit.javato hear the audio generated by Realtime API in real time.commit mode
Python
In the same directory as
tts_realtime_client.py, create another Python file namedcommit.pyand copy the following code into the file:import os import asyncio import logging import wave from tts_realtime_client import TTSRealtimeClient, SessionMode import pyaudio # QwenTTS service configuration # To use the instruction control feature, replace the model with qwen3-tts-instruct-flash-realtime and uncomment instructions in tts_realtime_client.py # 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/zh/model-studio/get-api-key # If you have not configured the environment variable, replace the next line with your Chinese 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()Run
commit.py. You can enter text to synthesize multiple times. Press Enter without entering text to hear the audio returned by Realtime API through the speaker.Java
In the same directory as
TTSRealtimeClient.java, create another Java file namedCommit.javaand copy the following code into the file: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/zh/model-studio/get-api-key // If you have not configured the environment variable, replace the next line with your Chinese 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) // To use instruction control, replace the model with qwen3-tts-instruct-flash-realtime and uncomment the instructions in TTSRealtimeClient.java TTSRealtimeClient client = new TTSRealtimeClient( URL, API_KEY, "Cherry", TTSRealtimeClient.SessionMode.COMMIT, audioData -> { playbackQueue.add(audioData); audioChunks.add(audioData); System.out.println("Received audio data: " + audioData.length + " bytes"); } ); client.connect(); // Interactive input System.out.println("Enter text (press Enter directly to send a commit event and end the session, press Ctrl+D to exit the program):"); Scanner scanner = new Scanner(System.in); while (true) { System.out.print("> "); if (!scanner.hasNextLine()) { client.finishSession(); break; } String userText = scanner.nextLine(); if (userText.isEmpty()) { // Empty input: commit the buffer and end the session System.out.println("Empty input, sending commit event and ending the session"); client.commitTextBuffer(); Thread.sleep(300); client.finishSession(); break; } else { System.out.println("Sending text: " + userText); client.appendText(userText); } } scanner.close(); // Wait for the response to complete client.waitForResponseDone(); client.waitForSessionFinished(); client.close(); // Wait for playback to complete playing.set(false); playerThread.join(); audioLine.drain(); audioLine.close(); // Save the audio file saveWav("output.wav"); System.out.println("Done"); } private static void saveWav(String filename) throws IOException { if (audioChunks.isEmpty()) { System.out.println("No audio data to save"); return; } ByteArrayOutputStream bos = new ByteArrayOutputStream(); for (byte[] chunk : audioChunks) { bos.write(chunk); } byte[] allAudio = bos.toByteArray(); AudioFormat format = new AudioFormat(SAMPLE_RATE, 16, 1, true, false); AudioInputStream ais = new AudioInputStream( new ByteArrayInputStream(allAudio), format, allAudio.length / 2); new File("outputs").mkdirs(); AudioSystem.write(ais, AudioFileFormat.Type.WAVE, new File("outputs/" + filename)); System.out.println("Audio saved to: outputs/" + filename); } }Compile and run
Commit.java. You can enter text to synthesize multiple times. Press Enter without entering text to hear the audio returned by Realtime API through the speaker. -
server_commit mode
Apply in production
Connection reuse (WebSocket)
WebSocket connections are reusable: after a synthesis task completes, you can start the next task on the same connection without re-establishing it.
Reuse process:
- Qwen-Audio-TTS / Qwen-Audio-TTS/CosyVoice: The client sends
finish-task, and after the server returnstask-finished, the client can sendrun-taskto start a new task. - Qwen-TTS: The client sends
session.finish, and after the server returnssession.finished, the client can create a new session to start the next task.
Reuse after cancellation: For Qwen-Audio-TTS / Qwen-Audio-TTS/CosyVoice, if you cancel the current task using the cancel directive, you can also send a new run-task on the same connection after the server returns task-finished. For details, see Cancel task.
Important
- Wait for the server to return the completion event (
task-finishedorsession.finished) before starting a new task. - Qwen-Audio-TTS, Qwen-Audio-TTS/CosyVoice require a different
task_idfor each task on a reused connection. - If a task fails, the server returns an error event and closes the connection. The connection can't be reused.
- If no new task is started within 60 seconds after the previous task ends, the connection is automatically closed.
For the event details of each model, see the corresponding API reference.
Rate limits
Model calls are subject to rate limits. When a limit is exceeded, the server returns the error Requests rate limit exceeded, please try again later. Reduce your request rate or concurrency and retry.
For the rate limits of each model, see Rate limiting.
High-concurrency best practices
The DashScope SDK has built-in pooling that reuses WebSocket connections and synthesizer objects, eliminating the overhead of repeatedly creating and destroying them.
View high-concurrency best practices
Qwen-Audio-TTS/CosyVoice
Qwen-Audio-TTS and Qwen-Audio-TTS/CosyVoice use the same SDK interface. The following examples also apply to Qwen-Audio-TTS models — simply replace the model and voice parameters.
Prerequisites
-
The DashScope SDK is installed and meets the version requirements. We recommend installing the latest version:
- Python SDK: version >= 1.25.2
- Java SDK: version >= 2.16.6
Python SDK
The Python SDK uses SpeechSynthesizerObjectPool to manage and reuse SpeechSynthesizer objects.
The pool creates a specified number of SpeechSynthesizer instances and establishes WebSocket connections at initialization. When you borrow an object, it's ready to send requests immediately, reducing first-packet latency. After the object is returned, the connection stays active for the next task.
Implementation steps
-
Install dependencies: Install the DashScope dependency (
pip install -U dashscope). -
Create and configure the object pool
Set the pool size to 1.5x-2x the peak concurrency, and don't exceed your account's QPS limit.
Create a global singleton pool (connection establishment during initialization takes some time):
from dashscope.audio.tts_v2 import SpeechSynthesizerObjectPool
synthesizer_object_pool = SpeechSynthesizerObjectPool(max_size=20)
import dashscope
# The following is the configuration for the China (Beijing) region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
dashscope.base_http_api_url = "https://{WorkspaceId}.cn-beijing.maas.aliyuncs.com/api/v1"
Important
- In the object pool scenario,
SpeechSynthesizerObjectPoolestablishes WebSocket connections with the server using the current globaldashscope.api_keyat initialization. The API key is written to theAuthorizationheader only during the WebSocket handshake for authentication. Subsequent task messages (such asrun-task) don't carry the API key. Modifyingdashscope.api_keyafter pool creation doesn't affect existing connections -- objects borrowed viaborrow_synthesizer(including those returned and re-borrowed) still use the API key from the original handshake. The new value is silently ignored, which may cause identity, quota, or billing attribution to differ from expectations. Note:borrow_synthesizerdoesn't support specifying an API key as a parameter. - To use multiple API keys, maintain a separate
SpeechSynthesizerObjectPoolinstance for each key.
-
Borrow a
SpeechSynthesizerobject from the poolIf the number of unreturned objects exceeds the pool capacity, the system creates additional objects.
These additional objects must establish new connections and don't benefit from pooling.
speech_synthesizer = connectionPool.borrow_synthesizer(
model='cosyvoice-v3-flash',
voice='longanyang',
seed=12382,
callback=synthesizer_callback
)
-
Perform speech synthesis
Call the
SpeechSynthesizerobject's call or streaming_call method to synthesize speech. -
Return the
SpeechSynthesizerobjectReturn the object after the task completes to make it available for reuse.
Don't return objects with incomplete or failed tasks.
connectionPool.return_synthesizer(speech_synthesizer)
Complete code
ImportantBefore using this code: SpeechSynthesizerObjectPool establishes WebSocket connections and authenticates using the current global dashscope.api_key at initialization. Modifying dashscope.api_key after pool creation doesn't affect existing connections -- the new value is silently ignored. For multiple API keys, maintain a separate pool instance for each key. For details, see the important note above.
# !/usr/bin/env python3
# Copyright (C) Alibaba Group. All Rights Reserved.
# MIT License (https://opensource.org/licenses/MIT)
import os
import time
import threading
import dashscope
from dashscope.audio.tts_v2 import *
USE_CONNECTION_POOL = True
text_to_synthesize = [
'Sentence 1: Welcome to Alibaba speech synthesis service.',
'Sentence 2: Welcome to Alibaba speech synthesis service.',
'Sentence 3: Welcome to Alibaba speech synthesis service.',
]
connectionPool = None
def init_dashscope_api_key():
'''
Set your DashScope API-key. More information:
https://github.com/aliyun/alibabacloud-bailian-speech-demo/blob/master/PREREQUISITES.md
'''
# The API Key differs between the Singapore and Beijing regions. Get your 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
def synthesis_text_to_speech_and_play_by_streaming_mode(text, task_id):
global USE_CONNECTION_POOL, connectionPool
'''
Synthesize speech with given text by streaming mode, async call and play the synthesized audio in real-time.
for more information, please refer to https://www.alibabacloud.com/help/document_detail/2712523.html
'''
complete_event = threading.Event()
# Define a callback to handle the result
class Callback(ResultCallback):
def on_open(self):
# when using object pool, on_open will be called after task start
self.file = open(f'result_{task_id}.mp3', 'wb')
print(f'[task_{task_id}] start')
def on_complete(self):
print(f'[task_{task_id}] speech synthesis task complete successfully.')
complete_event.set()
def on_error(self, message: str):
print(f'[task_{task_id}] speech synthesis task failed, {message}')
def on_close(self):
# when using object pool, on_open will be called after task finished
print(f'[task_{task_id}] finished')
def on_event(self, message):
# print(f'recv speech synthsis message {message}')
pass
def on_data(self, data: bytes) -> None:
# send to player
# save audio to file
self.file.write(data)
# Call the speech synthesizer callback
synthesizer_callback = Callback()
# Initialize the speech synthesizer
# you can customize the synthesis parameters, like voice, format, sample_rate or other parameters
if USE_CONNECTION_POOL:
speech_synthesizer = connectionPool.borrow_synthesizer(
model='cosyvoice-v3-flash',
voice='longanyang',
seed=12382,
callback=synthesizer_callback
)
else:
speech_synthesizer = SpeechSynthesizer(model='cosyvoice-v3-flash',
voice='longanyang',
seed=12382,
callback=synthesizer_callback)
try:
speech_synthesizer.call(text)
except Exception as e:
print(f'[task_{task_id}] speech synthesis task failed, {e}')
if USE_CONNECTION_POOL:
# close the synthesizer connection manually if task failed when using connection pool.
speech_synthesizer.close()
return
print('[task_{}] Synthesized text: {}'.format(task_id, text))
complete_event.wait()
print('[task_{}][Metric] requestId: {}, first package delay ms: {}'.format(
task_id,
speech_synthesizer.get_last_request_id(),
speech_synthesizer.get_first_package_delay()))
if USE_CONNECTION_POOL:
connectionPool.return_synthesizer(speech_synthesizer)
# main function
if __name__ == '__main__':
# You must set dashscope.api_key and base_websocket_api_url before creating SpeechSynthesizerObjectPool.
# The pool establishes WebSocket connections using the current global dashscope.api_key at initialization time.
# Modifying dashscope.api_key after pool creation will not affect existing connections in the pool.
# The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
dashscope.base_websocket_api_url='wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference'
init_dashscope_api_key()
if USE_CONNECTION_POOL:
print('creating connection pool')
start_time = time.time() * 1000
connectionPool = SpeechSynthesizerObjectPool(max_size=3)
end_time = time.time() * 1000
print('connection pool created, cost: {} ms'.format(end_time - start_time))
task_thread_list = []
for task_id in range(3):
thread = threading.Thread(
target=synthesis_text_to_speech_and_play_by_streaming_mode,
args=(text_to_synthesize[task_id], task_id))
task_thread_list.append(thread)
for task_thread in task_thread_list:
task_thread.start()
for task_thread in task_thread_list:
task_thread.join()
if USE_CONNECTION_POOL:
connectionPool.shutdown()
Resource management and error handling
-
Task succeeded: After a synthesis task completes normally, call
connectionPool.return_synthesizer(speech_synthesizer)to return theSpeechSynthesizerobject to the pool for reuse.ImportantDon't return
SpeechSynthesizerobjects with incomplete or failed tasks. -
Task failed: If an SDK internal error or business logic exception causes the task to abort, close the underlying WebSocket connection:
speech_synthesizer.close() -
After all synthesis tasks are complete, shut down the pool:
connectionPool.shutdown() -
When a TaskFailed error occurs on the server side, no additional handling is required.
Java SDK
The Java SDK achieves optimal performance through the coordination of a built-in connection pool and a custom object pool.
- Connection pool: The OkHttp3 connection pool integrated into the SDK manages and reuses underlying WebSocket connections, reducing network handshake overhead. This is enabled by default.
- Object pool: Built on
commons-pool2, this pool maintains a set of pre-connectedSpeechSynthesizerobjects. Borrowing from the pool eliminates connection setup latency, significantly reducing first-packet delay.
Implementation steps
-
Add dependencies
Add dashscope-sdk-java and commons-pool2 to your project's dependency configuration based on the build tool you use.
Maven and Gradle examples:
Maven
- Open the
pom.xmlfile of your Maven project. - Add the following dependencies inside the
<dependencies>tag.
<dependency> <groupId>com.alibaba</groupId> <artifactId>dashscope-sdk-java</artifactId> <!-- Replace 'the-latest-version' with version 2.16.9 or later. Check available versions at: https://mvnrepository.com/artifact/com.alibaba/dashscope-sdk-java --> <version>the-latest-version</version> </dependency> <dependency> <groupId>org.apache.commons</groupId> <artifactId>commons-pool2</artifactId> <!-- Replace 'the-latest-version' with the latest version. Check available versions at: https://mvnrepository.com/artifact/org.apache.commons/commons-pool2 --> <version>the-latest-version</version> </dependency>- Save the
pom.xmlfile. - Run a Maven command such as
mvn clean installormvn compileto update the project dependencies.
Gradle
- Open the
build.gradlefile of your Gradle project. - Add the following dependencies inside the
dependenciesblock.
dependencies { // Replace 'the-latest-version' with version 2.16.6 or later. Check available versions at: https://mvnrepository.com/artifact/com.alibaba/dashscope-sdk-java implementation group: 'com.alibaba', name: 'dashscope-sdk-java', version: 'the-latest-version' // Replace 'the-latest-version' with the latest version. Check available versions at: https://mvnrepository.com/artifact/org.apache.commons/commons-pool2 implementation group: 'org.apache.commons', name: 'commons-pool2', version: 'the-latest-version' }- Save the
build.gradlefile. - In the terminal, navigate to the project root directory and run the following Gradle command to update the dependencies.
./gradlew build --refresh-dependenciesOn Windows, use the following command instead:
gradlew build --refresh-dependencies - Open the
-
Configure the connection pool
Configure the key connection pool parameters through environment variables:
Environment variable
Description
DASHSCOPE_CONNECTION_POOL_SIZE
Connection pool size.
Recommended value: at least 2x the peak concurrency.
Default: 32.
DASHSCOPE_MAXIMUM_ASYNC_REQUESTS
Maximum number of async requests.
Recommended value: same as
DASHSCOPE_CONNECTION_POOL_SIZE.Default: 32.
DASHSCOPE_MAXIMUM_ASYNC_REQUESTS_PER_HOST
Maximum number of async requests per host.
Recommended value: same as
DASHSCOPE_CONNECTION_POOL_SIZE.Default: 32.
-
Configure the object pool
Configure the object pool size through environment variables:
Environment variable
Description
COSYVOICE_OBJECTPOOL_SIZE
Object pool size.
Recommended value: 1.5x-2x the peak concurrency.
Default: 500.
Important
- The object pool size (
COSYVOICE_OBJECTPOOL_SIZE) must be less than or equal to the connection pool size (DASHSCOPE_CONNECTION_POOL_SIZE). Otherwise, when the object pool requests objects and the connection pool is full, calling threads block while waiting for available connections. - The object pool size must not exceed your account's QPS (queries per second) limit.
Create the object pool with the following code:
- The object pool size (
class CosyvoiceObjectPool {
// ... Other code is omitted here. For the complete example, see the full code.
public static GenericObjectPool<SpeechSynthesizer> getInstance() {
lock.lock();
if (synthesizerPool == null) {
// You can set the object pool size here, or set it in the environment variable COSYVOICE_OBJECTPOOL_SIZE.
// It is recommended to set it to 1.5 to 2 times the maximum concurrent connections of the server.
int objectPoolSize = getObjectivePoolSize();
SpeechSynthesizerObjectFactory speechSynthesizerObjectFactory =
new SpeechSynthesizerObjectFactory();
GenericObjectPoolConfig<SpeechSynthesizer> config =
new GenericObjectPoolConfig<>();
config.setMaxTotal(objectPoolSize);
config.setMaxIdle(objectPoolSize);
config.setMinIdle(objectPoolSize);
synthesizerPool =
new GenericObjectPool<>(speechSynthesizerObjectFactory, config);
}
lock.unlock();
return synthesizerPool;
}
}
-
Borrow a
SpeechSynthesizerobject from the poolIf the number of unreturned objects exceeds the pool's maximum capacity, the system creates an additional
SpeechSynthesizerobject.These newly created objects require re-initialization and a new WebSocket connection, so they don't benefit from pooling.
synthesizer = CosyvoiceObjectPool.getInstance().borrowObject();
-
Perform speech synthesis
After borrowing a
SpeechSynthesizerobject from the pool, callupdateParamAndCallback(param, callback)to bind the parameters and callback for the current task, then callstreamingCallorcallto synthesize speech.Important
- In the object pool scenario,
updateParamAndCallbackis called multiple times (once each time an object is borrowed, to set the callback and task-level parameters such asvoiceandformat). TheapiKeypassed in each call must remain the same.updateParamAndCallbackonly updates the local fields of the currentSpeechSynthesizerinstance and doesn't rebuild the underlying WebSocket connection. The SDK writes theapiKeyinto theAuthorizationheader only during the WebSocket handshake for authentication. Subsequent task messages (such asrun-task) don't carry theapiKey. As long as the reused connection isn't disconnected, passing a newapiKeywon't be sent to the server -- the request still uses theapiKeyfrom the initial handshake, which may cause identity, quota, or billing attribution to differ from expectations. - To use multiple API keys, maintain separate object pool instances for each key.
- In the object pool scenario,
-
Return the
SpeechSynthesizerobjectAfter the synthesis task completes, return the
SpeechSynthesizerobject so subsequent tasks can reuse it.Don't return objects with incomplete or failed tasks.
CosyvoiceObjectPool.getInstance().returnObject(synthesizer);
Complete code
ImportantBefore using this code: In the object pool scenario, the apiKey passed to updateParamAndCallback across multiple calls must remain the same -- the SDK doesn't update the API key of established connections, and passing a different API key has no effect. For multiple API keys, maintain a separate object pool instance for each key. For details, see the important note above.
import com.alibaba.dashscope.audio.tts.SpeechSynthesisResult;
import com.alibaba.dashscope.audio.ttsv2.SpeechSynthesisAudioFormat;
import com.alibaba.dashscope.audio.ttsv2.SpeechSynthesisParam;
import com.alibaba.dashscope.audio.ttsv2.SpeechSynthesizer;
import com.alibaba.dashscope.common.ResultCallback;
import com.alibaba.dashscope.exception.NoApiKeyException;
import com.alibaba.dashscope.utils.Constants;
import lombok.extern.slf4j.Slf4j;
import org.apache.commons.pool2.BasePooledObjectFactory;
import org.apache.commons.pool2.PooledObject;
import org.apache.commons.pool2.impl.DefaultPooledObject;
import org.apache.commons.pool2.impl.GenericObjectPool;
import org.apache.commons.pool2.impl.GenericObjectPoolConfig;
import java.time.LocalDateTime;
import java.util.concurrent.ExecutorService;
import java.util.concurrent.Executors;
import java.util.concurrent.TimeUnit;
import java.util.concurrent.locks.Lock;
/**
* You need to include the org.apache.commons.pool2 and DashScope packages in your project.
*
* DashScope SDK 2.16.6 and later versions are optimized for high-concurrency scenarios.
* Versions earlier than DashScope SDK 2.16.6 are not recommended for high-concurrency use.
*
*
* Before making high-concurrency calls to the TTS service,
* please configure the connection pool parameters through the following environment variables.
*
* DASHSCOPE_MAXIMUM_ASYNC_REQUESTS
* DASHSCOPE_MAXIMUM_ASYNC_REQUESTS_PER_HOST
* DASHSCOPE_CONNECTION_POOL_SIZE
*
*/
class SpeechSynthesizerObjectFactory
extends BasePooledObjectFactory<SpeechSynthesizer> {
public SpeechSynthesizerObjectFactory() {
super();
}
@Override
public SpeechSynthesizer create() throws Exception {
return new SpeechSynthesizer();
}
@Override
public PooledObject<SpeechSynthesizer> wrap(SpeechSynthesizer obj) {
return new DefaultPooledObject<>(obj);
}
}
class CosyvoiceObjectPool {
public static GenericObjectPool<SpeechSynthesizer> synthesizerPool;
public static String COSYVOICE_OBJECTPOOL_SIZE_ENV = "COSYVOICE_OBJECTPOOL_SIZE";
public static int DEFAULT_OBJECT_POOL_SIZE = 500;
private static Lock lock = new java.util.concurrent.locks.ReentrantLock();
public static int getObjectivePoolSize() {
try {
Integer n = Integer.parseInt(System.getenv(COSYVOICE_OBJECTPOOL_SIZE_ENV));
System.out.println("Using Object Pool Size In Env: "+ n);
return n;
} catch (NumberFormatException e) {
System.out.println("Using Default Object Pool Size: "+ DEFAULT_OBJECT_POOL_SIZE);
return DEFAULT_OBJECT_POOL_SIZE;
}
}
public static GenericObjectPool<SpeechSynthesizer> getInstance() {
lock.lock();
if (synthesizerPool == null) {
// You can set the object pool size here or in the COSYVOICE_OBJECTPOOL_SIZE environment variable.
// It is recommended to set it to 1.5 to 2 times your server's maximum concurrent connections.
int objectPoolSize = getObjectivePoolSize();
SpeechSynthesizerObjectFactory speechSynthesizerObjectFactory =
new SpeechSynthesizerObjectFactory();
GenericObjectPoolConfig<SpeechSynthesizer> config =
new GenericObjectPoolConfig<>();
config.setMaxTotal(objectPoolSize);
config.setMaxIdle(objectPoolSize);
config.setMinIdle(objectPoolSize);
synthesizerPool =
new GenericObjectPool<>(speechSynthesizerObjectFactory, config);
}
lock.unlock();
return synthesizerPool;
}
}
class SynthesizeTaskWithCallback implements Runnable {
String[] textArray;
String requestId;
long timeCost;
public SynthesizeTaskWithCallback(String[] textArray) {
this.textArray = textArray;
}
@Override
public void run() {
SpeechSynthesizer synthesizer = null;
long startTime = System.currentTimeMillis();
// if recv onError
final boolean[] hasError = {false};
try {
class ReactCallback extends ResultCallback<SpeechSynthesisResult> {
ReactCallback() {}
@Override
public void onEvent(SpeechSynthesisResult message) {
if (message.getAudioFrame() != null) {
try {
byte[] bytesArray = message.getAudioFrame().array();
System.out.println("Received audio, audio stream length: " + bytesArray.length);
} catch (Exception e) {
throw new RuntimeException(e);
}
}
}
@Override
public void onComplete() {}
@Override
public void onError(Exception e) {
System.out.println(e.getMessage());
e.printStackTrace();
hasError[0] = true;
}
}
SpeechSynthesisParam param =
SpeechSynthesisParam.builder()
.model("cosyvoice-v3-flash")
.voice("longanyang")
// The API Key differs between the Singapore and Beijing regions. Get your API Key: https://www.alibabacloud.com/help/zh/model-studio/get-api-key
// If you have not configured the environment variable, replace the next line with your Chinese Model Studio API Key: .apiKey("sk-xxx")
.apiKey(System.getenv("DASHSCOPE_API_KEY"))
.format(SpeechSynthesisAudioFormat
.MP3_22050HZ_MONO_256KBPS) // Use PCM or MP3 for streaming synthesis
.build();
try {
synthesizer = CosyvoiceObjectPool.getInstance().borrowObject();
// Note: In an object pool scenario, the apiKey passed in multiple calls to updateParamAndCallback must always be the same. The SDK does not update the apiKey for an established connection, so passing a different apiKey will not take effect. See the important notes in the "Perform speech synthesis" step above.
synthesizer.updateParamAndCallback(param, new ReactCallback());
for (String text : textArray) {
synthesizer.streamingCall(text);
}
Thread.sleep(20);
synthesizer.streamingComplete(60000);
requestId = synthesizer.getLastRequestId();
} catch (Exception e) {
System.out.println("Exception e: " + e.toString());
hasError[0] = true;
}
} catch (Exception e) {
hasError[0] = true;
throw new RuntimeException(e);
}
if (synthesizer != null) {
try {
if (hasError[0] == true) {
// If an exception occurs, close the connection and invalidate the object in the pool.
synthesizer.getDuplexApi().close(1000, "bye");
CosyvoiceObjectPool.getInstance().invalidateObject(synthesizer);
} else {
// If the task completes normally, return the object to the pool.
CosyvoiceObjectPool.getInstance().returnObject(synthesizer);
}
} catch (Exception e) {
throw new RuntimeException(e);
}
long endTime = System.currentTimeMillis();
timeCost = endTime - startTime;
System.out.println("[Thread " + Thread.currentThread() + "] Speech synthesis task completed. Time cost: " + timeCost + " ms, RequestId " + requestId);
}
}
}
@Slf4j
public class SynthesizeTextToSpeechWithCallbackConcurrently {
public static void checkoutEnv(String envName, int defaultSize) {
if (System.getenv(envName) != null) {
System.out.println("[ENV CHECK]: " + envName + " "
+ System.getenv(envName));
} else {
System.out.println("[ENV CHECK]: " + envName
+ " Using Default which is " + defaultSize);
}
}
public static void main(String[] args)
throws InterruptedException, NoApiKeyException {
// The following is the configuration for the Singapore region. Replace "{WorkspaceId}" with your actual workspace ID. The configuration varies by region.
Constants.baseWebsocketApiUrl = "wss://{WorkspaceId}.ap-southeast-1.maas.aliyuncs.com/api-ws/v1/inference";
// Check for connection pool env
checkoutEnv("DASHSCOPE_CONNECTION_POOL_SIZE", 32);
checkoutEnv("DASHSCOPE_MAXIMUM_ASYNC_REQUESTS", 32);
checkoutEnv("DASHSCOPE_MAXIMUM_ASYNC_REQUESTS_PER_HOST", 32);
checkoutEnv(CosyvoiceObjectPool.COSYVOICE_OBJECTPOOL_SIZE_ENV, CosyvoiceObjectPool.DEFAULT_OBJECT_POOL_SIZE);
int runTimes = 3;
// Create the pool of SpeechSynthesis objects
ExecutorService executorService = Executors.newFixedThreadPool(runTimes);
for (int i = 0; i < runTimes; i++) {
// Record the task submission time
LocalDateTime submissionTime = LocalDateTime.now();
executorService.submit(new SynthesizeTaskWithCallback(new String[] {
"Before my bed, moonlight shines bright,", "I wonder if it is frost on the ground,", "I raise my eyes to gaze at the bright moon,", "then bow my head, thinking of home."}));
}
// Shut down the ExecutorService and wait for all tasks to complete
executorService.shutdown();
executorService.awaitTermination(1, TimeUnit.MINUTES);
System.exit(0);
}
}
Recommended configuration
The following configuration is based on test results from running only the Qwen-Audio-TTS/CosyVoice speech synthesis service on Alibaba Cloud ECS instances with the specified specifications. Excessive concurrency may increase task processing latency.
The "single-machine concurrency" refers to the number of Qwen-Audio-TTS/CosyVoice synthesis tasks running simultaneously, equivalent to the number of worker threads.
ECS specifications (Alibaba Cloud) | Max single-machine concurrency | Object pool size | Connection pool size |
|---|---|---|---|
4 vCPUs, 8 GiB | 100 | 500 | 2000 |
8 vCPUs, 16 GiB | 150 | 500 | 2000 |
16 vCPUs, 32 GiB | 200 | 500 | 2000 |
Resource management and error handling
-
Task succeeded: After a synthesis task completes normally, call the GenericObjectPool returnObject method to return the
SpeechSynthesizerobject to the pool for reuse.In the sample code, this corresponds to
CosyvoiceObjectPool.getInstance().returnObject(synthesizer)ImportantDon't return
SpeechSynthesizerobjects with incomplete or failed tasks. -
Task failed: If an SDK internal error or business logic exception causes the task to abort, perform the following:
- Close the underlying WebSocket connection.
- Invalidate the object in the pool to prevent it from being reused.
// In the current code, the corresponding content is as follows
// Close the connection
synthesizer.getDuplexApi().close(1000, "bye");
// Invalidate the synthesizer in the object pool when an exception occurs
CosyvoiceObjectPool.getInstance().invalidateObject(synthesizer);
- When a TaskFailed error occurs on the server side, no additional handling is required.
Warmup and latency measurement
When evaluating the DashScope Java SDK for concurrency latency and performance, run sufficient warmup operations first. This ensures that measurements reflect steady-state performance rather than initial connection overhead.
Connection reuse mechanism
The DashScope Java SDK uses a global singleton connection pool to efficiently manage and reuse WebSocket connections, reducing the overhead of frequent connection creation and teardown for high-concurrency workloads.
How this mechanism works:
-
On-demand creation: The SDK doesn't pre-create WebSocket connections at startup. Connections are established on the first call.
-
Time-limited reuse: After a request completes, the connection remains in the pool for up to 60 seconds for reuse.
- If a new request arrives within 60 seconds, the existing connection is reused, avoiding handshake overhead.
- If the connection sits idle for more than 60 seconds, it's automatically closed to free resources.
Importance of warmup
In the following scenarios, the connection pool may have no reusable active connections, forcing new connections to be created:
- The application just started and hasn't made any calls yet.
- The service has been idle for more than 60 seconds, and connections in the pool have timed out and closed.
In these scenarios, the first request must complete a full WebSocket handshake (TCP handshake, TLS negotiation, protocol upgrade), resulting in significantly higher latency than subsequent requests that reuse connections. Without warmup, performance test results are skewed by this initial connection overhead.
SDK-reported latency vs. actual first-packet latency
The first-packet latency reported by the SDK (for example, the value from get_first_package_delay()) includes WebSocket connection setup and network transmission time, and doesn't equal the model service's actual first-packet latency.
The actual first-packet latency is the interval between when the server receives the run-task instruction and when it returns the first result-generated event. This value can be found in server-side logs.
In high-concurrency scenarios, due to simultaneous connection creation and resource scheduling, the SDK-reported latency may be significantly higher than the actual server-side first-packet latency. If you observe high SDK-reported first-packet latency:
- Compare with the server-side log's first-packet latency (from
run-taskto the firstresult-generated) to check whether model inference performance is normal. - Use the object pool or connection pool mechanism described above for warmup. This eliminates WebSocket handshake overhead, bringing the SDK-reported latency closer to the actual first-packet latency.
Recommended practices
For reliable performance data, follow these warmup steps before running load tests or collecting latency metrics:
- Simulate the target concurrency level and send a number of warmup requests in advance (for example, sustain traffic for 1-2 minutes) to fully populate the connection pool.
- Confirm that the connection pool has established and maintained sufficient active connections before starting formal performance data collection.
Proper warmup brings the SDK connection pool to a stable reuse state, producing latency metrics that accurately reflect online steady-state performance.
Common Java SDK exceptions
Exception 1: Server TCP connections keep increasing despite stable traffic
Each SDK object creates a connection upon initialization. Without an object pool, the object is destroyed after each task completes. The connection then enters an unreferenced state and remains open until the server times it out after 61 seconds. During this period, the connection can't be reused.
Under high concurrency, new tasks that find no reusable connections create new ones, leading to:
- Continuously increasing connection count.
- Server resource exhaustion and degraded performance due to excessive connections.
- Connection pool saturation, causing new tasks to block while waiting for available connections.
The object pool's MaxIdle is configured lower than MaxTotal, causing idle objects beyond MaxIdle to be destroyed, which leaks their connections. These leaked connections must wait 61 seconds for timeout before disconnecting, same as Type 1.
Solution:
For Type 1: Use an object pool.
For Type 2: Check the object pool configuration and set MaxIdle equal to MaxTotal. Disable the automatic object eviction policy.
Exception 2: Task takes 60 seconds longer than normal
Same root cause as Exception 1: The connection pool has reached its maximum connection limit, and new tasks must wait 61 seconds for unreferenced connections to time out before a connection becomes available.
Exception 3: Tasks are slow at service startup but gradually recover
Root cause:
Under high concurrency, a single object reuses the same WebSocket connection. WebSocket connections are only created at service startup. If high concurrency starts immediately at launch, creating too many WebSocket connections simultaneously causes blocking.
Solution:
Gradually ramp up concurrency after starting the service, or add warmup tasks.
Exception 4: Server reports "Invalid action('run-task')! Please follow the protocol!"
Root cause:
A client-side error occurred, but the server didn't detect it. The connection remains in an in-task state. When this connection and object are reused for the next task, the protocol flow breaks and the next task fails.
Solution:
After an exception is thrown, close the WebSocket connection before returning the object to the pool.
Exception 5: Traffic spikes despite stable business load
Root cause:
Creating too many WebSocket connections simultaneously causes blocking. While blocked, incoming business traffic accumulates. Once the blockage clears, all accumulated tasks fire at once, creating a traffic spike that may exceed your account's concurrency limit, causing task failures or server degradation.
Common triggers for simultaneously creating too many WebSocket connections:
- Service startup phase
- Network anomalies causing many WebSocket connections to disconnect and reconnect simultaneously
- Burst of server-side errors causing mass WebSocket reconnections. Common errors include exceeding the account concurrency limit ("Requests rate limit exceeded, please try again later.").
Solution:
- Check network conditions.
- Investigate whether a burst of other server-side errors preceded the spike.
- Increase the account concurrency limit.
- Reduce the object pool and connection pool sizes to cap maximum concurrency through pool limits.
- Upgrade server specifications or add more machines.
Exception 6: All tasks slow down as concurrency increases
Solution:
- Check whether network bandwidth has reached its limit.
- Check whether the actual concurrency is too high.
Supported models and regions
Singapore
To call the following models, select an API Key from the Singapore region:
-
Qwen-Audio-TTS: qwen-audio-3.0-tts-plus, qwen-audio-3.0-tts-flash
-
Qwen-Audio-TTS/CosyVoice: cosyvoice-v3-plus, cosyvoice-v3-flash
-
Qwen-TTS:
- Qwen3-TTS-Instruct-Flash-Realtime: qwen3-tts-instruct-flash-realtime (stable, currently equivalent to qwen3-tts-instruct-flash-realtime-2026-01-22), qwen3-tts-instruct-flash-realtime-2026-01-22 (latest snapshot)
- Qwen3-TTS-VD-Realtime: qwen3-tts-vd-realtime-2026-01-15 (latest snapshot), qwen3-tts-vd-realtime-2025-12-16 (snapshot)
- Qwen3-TTS-VC-Realtime: qwen3-tts-vc-realtime-2026-01-15 (latest snapshot), qwen3-tts-vc-realtime-2025-11-27 (snapshot)
- Qwen3-TTS-Flash-Realtime: qwen3-tts-flash-realtime (stable, currently equivalent to qwen3-tts-flash-realtime-2025-11-27), qwen3-tts-flash-realtime-2025-11-27 (latest snapshot), qwen3-tts-flash-realtime-2025-09-18 (snapshot)
China (Beijing)
To call the following models, select an API Key from the Beijing region:
-
Qwen-Audio-TTS: qwen-audio-3.0-tts-plus, qwen-audio-3.0-tts-flash
-
Qwen-Audio-TTS/CosyVoice: cosyvoice-v3.5-plus, cosyvoice-v3.5-flash, cosyvoice-v3-plus, cosyvoice-v3-flash, cosyvoice-v2
-
Qwen-TTS:
- Qwen3-TTS-Instruct-Flash-Realtime: qwen3-tts-instruct-flash-realtime (stable, currently equivalent to qwen3-tts-instruct-flash-realtime-2026-01-22), qwen3-tts-instruct-flash-realtime-2026-01-22 (latest snapshot)
- Qwen3-TTS-VD-Realtime: qwen3-tts-vd-realtime-2026-01-15 (latest snapshot), qwen3-tts-vd-realtime-2025-12-16 (snapshot)
- Qwen3-TTS-VC-Realtime: qwen3-tts-vc-realtime-2026-01-15 (latest snapshot), qwen3-tts-vc-realtime-2025-11-27 (snapshot)
- Qwen3-TTS-Flash-Realtime: qwen3-tts-flash-realtime (stable, currently equivalent to qwen3-tts-flash-realtime-2025-11-27), qwen3-tts-flash-realtime-2025-11-27 (latest snapshot), qwen3-tts-flash-realtime-2025-09-18 (snapshot)
- Qwen-TTS-Realtime: qwen-tts-realtime (stable, currently equivalent to qwen-tts-realtime-2025-07-15), qwen-tts-realtime-latest (latest, currently equivalent to qwen-tts-realtime-2025-07-15), qwen-tts-realtime-2025-07-15 (snapshot)
Supported voices
Different models support different voices. Set the voice request parameter to the value in the voice parameter column of the corresponding voice list.
API reference
- Real-time speech synthesis - Qwen-Audio-TTS/CosyVoice API reference
- Real-time speech synthesis - Qwen-TTS API reference
- AOQ client SDK (for CosyVoice models)
FAQ
Q: How do I fix incorrect pronunciation in speech synthesis? How do I control the pronunciation of polyphonic characters?
- Replace the polyphonic character with a homophone to quickly fix the pronunciation issue.
- Use SSML markup to control pronunciation .
Q: How do I troubleshoot silent audio when using a cloned voice?
-
Verify the voice status
Call the Voice cloning/design API interface and confirm that the voice
statusisOK. -
Check model version consistency
Make sure the
target_modelparameter used during voice cloning matches themodelparameter used for speech synthesis. For example:- Cloning used
cosyvoice-v3-plus - Synthesis must also use
cosyvoice-v3-plus
- Cloning used
-
Verify the source audio quality
Check whether the source audio used for voice cloning meets the requirements in Voice cloning/design API:
- Audio duration: 10-20 seconds
- Clear audio quality
- No background noise
-
Check the request parameters
Confirm that the
voiceparameter in the speech synthesis request is set to the cloned voice ID.
Q: What should I do if the synthesized audio from a cloned voice is unstable or incomplete?
If the synthesized audio from a cloned voice has any of the following issues:
- Audio playback is incomplete, with only part of the text spoken
- Synthesis quality is inconsistent
- Audio contains abnormal pauses or silent segments
Possible cause: The source audio doesn't meet the quality requirements.
Solution: Check whether the source audio meets the requirements in Recording guide for voice cloning. Re-record the audio following the recording guidelines.
Q: Why does the actual duration of synthesized audio differ from the duration shown in the WAV file?
Speech synthesis uses a streaming mechanism that returns data as it's generated. The duration in the saved WAV file header is an estimate and may be inaccurate. For precise duration, set the format to pcm, wait for the full synthesis result, and then add the WAV file header manually.
Q: Why can't the audio file play?
Troubleshoot based on your scenario:
-
Audio saved as a complete file (such as xx.mp3)
- Audio format consistency: The audio format in the request parameters must match the file extension (for example, if the parameter is wav, the file must be .wav).
- Player compatibility: Confirm that the player supports the audio format and sample rate.
-
Streaming audio playback
- Save the audio stream as a complete file and try playing it with a media player. If the file doesn't play, refer to scenario 1 above.
- If the file plays correctly, the issue is in the streaming playback implementation. Confirm that the player supports streaming playback (such as ffmpeg, pyaudio, AudioFormat, or MediaSource).
Q: Why is audio playback stuttering?
Troubleshoot with the following steps:
-
Check the text sending rate: Make sure the sending interval is reasonable to avoid the previous audio segment finishing before the next text arrives.
-
Check callback function performance:
- Confirm that no blocking logic exists in the callback function.
- Callbacks run on the WebSocket thread. Blocking operations affect data reception. Write audio data to a separate buffer and process it in another thread.
-
Check network stability: Network fluctuations can cause audio transmission interruptions or delays.
Q: Why is speech synthesis taking a long time?
Troubleshoot with the following steps:
-
Check input intervals
For streaming synthesis, check whether the text sending interval is too long. Long intervals increase the total synthesis time.
-
Analyze performance metrics
- First-packet latency: normally around 500 ms.
- RTF (Real-Time Factor = total synthesis time / audio duration): should be less than 1.0.
Q: How do I restrict an API key to only the speech synthesis service (permission isolation)?
Create a new workspace and grant access to specific models only. This limits the scope of the API key. For details, see Manage workspaces.