Signed-off-by: lishunyang <lishunyang12@163.com> Signed-off-by: reidliu41 <reid201711@gmail.com> Signed-off-by: Alex Brooks <albrooks@redhat.com> Co-authored-by: reidliu41 <reid201711@gmail.com> Co-authored-by: xiaohajiayou <75477391+xiaohajiayou@users.noreply.github.com> Co-authored-by: Alex Brooks <albrooks@redhat.com> Co-authored-by: Hongsheng Liu <liuhongsheng4@huawei.com>
17 KiB
Qwen3-TTS
Source https://github.com/vllm-project/vllm-omni/tree/main/examples/online_serving/qwen3_tts.
🛠️ Installation
Please refer to README.md
Supported Models
| Model | Task Type | Description |
|---|---|---|
Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice |
CustomVoice | Predefined speaker voices with optional style control |
Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign |
VoiceDesign | Natural language voice style description |
Qwen/Qwen3-TTS-12Hz-1.7B-Base |
Base | Voice cloning from reference audio |
Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice |
CustomVoice | Smaller/faster variant |
Qwen/Qwen3-TTS-12Hz-0.6B-Base |
Base | Smaller/faster variant for voice cloning |
Gradio Demo
!!! note "Gradio is an optional dependency"
The Gradio demo requires the [demo] extras. Install them first:
```bash
pip install 'vllm-omni[demo]'
```
Or, if installing from source: `pip install -e '.[demo]'`
Two interactive Gradio demos are available, both supporting all 3 task types:
| Demo | File | Transport | Streaming Quality |
|---|---|---|---|
| Standard | gradio_demo.py |
HTTP chunked | May have small gaps between chunks |
| FastRTC | gradio_fastrtc_demo.py |
WebRTC | Gapless streaming (requires pip install fastrtc) |
# Option 1: Launch server + Standard Gradio together
./run_gradio_demo.sh # CustomVoice (default)
./run_gradio_demo.sh --task-type VoiceDesign # VoiceDesign
./run_gradio_demo.sh --task-type Base # Voice cloning
# Option 2: If server is already running
python gradio_demo.py --api-base http://localhost:8000
# Option 3: FastRTC demo (gapless streaming)
pip install fastrtc
python gradio_fastrtc_demo.py --api-base http://localhost:8000
Then open http://localhost:7860 in your browser.
Run examples (Qwen3-TTS)
Launch the Server
# CustomVoice model (predefined speakers)
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
--deploy-config vllm_omni/deploy/qwen3_tts.yaml \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager
# VoiceDesign model
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign \
--deploy-config vllm_omni/deploy/qwen3_tts.yaml \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager
# Base model (voice cloning)
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-Base \
--deploy-config vllm_omni/deploy/qwen3_tts.yaml \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager
If you have custom stage configs file, launch the server with command below
vllm serve Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
--stage-configs-path /path/to/stage_configs_file \
--omni \
--port 8091 \
--trust-remote-code \
--enforce-eager
Alternatively, use the convenience script:
./run_server.sh # Default: CustomVoice model
./run_server.sh CustomVoice # CustomVoice model
./run_server.sh VoiceDesign # VoiceDesign model
./run_server.sh Base # Base (voice clone) model
Send TTS Request
Get into the example folder
cd examples/online_serving/qwen3_tts
Send request via python
# CustomVoice: Use predefined speaker
python openai_speech_client.py \
--text "你好,我是通义千问" \
--speaker vivian \
--language Chinese
# CustomVoice with style instruction
python openai_speech_client.py \
--text "今天天气真好" \
--speaker ryan \
--instructions "用开心的语气说"
# VoiceDesign: Describe the voice style
python openai_speech_client.py \
--model Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign \
--task-type VoiceDesign \
--text "哥哥,你回来啦" \
--instructions "体现撒娇稚嫩的萝莉女声,音调偏高"
# Base: Voice cloning
python openai_speech_client.py \
--model Qwen/Qwen3-TTS-12Hz-1.7B-Base \
--task-type Base \
--text "Hello, this is a cloned voice" \
--ref-audio /path/to/reference.wav \
--ref-text "Original transcript of the reference audio"
The Python client supports the following command-line arguments:
--api-base: API base URL (default:http://localhost:8091)--model(or-m): Model name/path (default:Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice)--task-type(or-t): TTS task type. Options:CustomVoice,VoiceDesign,Base--text: Text to synthesize (required)--speaker: Speaker name (default:vivian). Options:vivian,ryan,aiden, etc.--language: Language. Options:Auto,Chinese,English,Japanese,Korean,German,French,Russian,Portuguese,Spanish,Italian--instructions: Voice style/emotion instructions--ref-audio: Reference audio file path or URL for voice cloning (Base task). Local paths are automatically base64-encoded by the client before sending to the server.--ref-text: Reference audio transcript for voice cloning (Base task).--response-format: Audio output format (default:wav). Options:wav,mp3,flac,pcm,aac,opus--output(or-o): Output audio file path (default:tts_output.wav)
Send request via curl
# Simple TTS request
curl -X POST http://localhost:8091/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{
"input": "Hello, how are you?",
"voice": "vivian",
"language": "English"
}' --output output.wav
# With style instruction
curl -X POST http://localhost:8091/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{
"input": "I am so excited!",
"voice": "vivian",
"instructions": "Speak with great enthusiasm"
}' --output excited.wav
# List available voices in CustomVoice models
curl http://localhost:8091/v1/audio/voices
Using OpenAI SDK
from openai import OpenAI
client = OpenAI(base_url="http://localhost:8091/v1", api_key="none")
response = client.audio.speech.create(
model="Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice",
voice="vivian",
input="Hello, how are you?",
)
response.stream_to_file("output.wav")
Using Python httpx
import httpx
response = httpx.post(
"http://localhost:8091/v1/audio/speech",
json={
"input": "Hello, how are you?",
"voice": "vivian",
"language": "English",
},
timeout=300.0,
)
with open("output.wav", "wb") as f:
f.write(response.content)
API Reference
Voices Endpoint
GET /v1/audio/voices
List all available voices/speakers from the loaded model, including both built-in model voices and uploaded custom voices.
Response Example:
{
"voices": ["vivian", "ryan", "custom_voice_1"],
"uploaded_voices": [
{
"name": "custom_voice_1",
"consent": "user_consent_id",
"created_at": 1738660000,
"file_size": 1024000,
"mime_type": "audio/wav",
"ref_text": "The exact transcript of the audio sample.",
"speaker_description": "warm narrator"
}
]
}
Fields ref_text and speaker_description are omitted per-entry when not provided at upload time.
POST /v1/audio/voices
Upload a new voice sample for voice cloning in Base task TTS requests.
Form Parameters:
audio_sample(required): Audio file (max 10MB, supported formats: wav, mp3, flac, ogg, aac, webm, mp4)consent(required): Consent recording IDname(required): Name for the new voiceref_text(optional): Transcript of the audio. Enables in-context voice cloning (higher quality).speaker_description(optional): Free-form description of the voice (e.g. "warm narrator", "energetic presenter"). Stored as metadata.
Response Example:
{
"success": true,
"voice": {
"name": "custom_voice_1",
"consent": "user_consent_id",
"created_at": 1738660000,
"mime_type": "audio/wav",
"file_size": 1024000,
"ref_text": "The exact transcript of the audio sample.",
"speaker_description": "warm narrator"
}
}
Fields ref_text and speaker_description are omitted when not provided at upload time.
Usage Example:
curl -X POST http://localhost:8091/v1/audio/voices \
-F "audio_sample=@/path/to/voice_sample.wav" \
-F "consent=user_consent_id" \
-F "name=custom_voice_1" \
-F "ref_text=The exact transcript of the audio sample." \
-F "speaker_description=warm narrator"
Endpoint
POST /v1/audio/speech
Content-Type: application/json
This endpoint follows the OpenAI Audio Speech API format with additional Qwen3-TTS parameters.
Request Body
{
"input": "Text to synthesize",
"voice": "vivian",
"response_format": "wav",
"task_type": "CustomVoice",
"language": "Auto",
"instructions": "Optional style instructions",
"ref_audio": "HTTP URL, base64 data URL, or file:// URI for voice cloning",
"ref_text": "Reference audio transcript",
"x_vector_only_mode": false,
"max_new_tokens": 2048
}
Note: The
modelfield is optional when serving a single model, as the server already knows which model is loaded.
Response
Returns binary audio data with appropriate Content-Type header (e.g., audio/wav).
Voice and language (summary)
- Speaker: Use the
voicerequest field to select the speaker (e.g.,vivian,ryan,aiden). List available speakers withGET /v1/audio/voices. - Language: Use the
languagefield for the codec language tag (Auto,Chinese,English, etc.). Default isAutofor automatic detection. - CustomVoice: Requires a valid
voicefrom the model’s speaker set. VoiceDesign: Useinstructionsto describe the voice. Base: Useref_audioandref_textfor voice cloning.
Parameters
OpenAI Standard Parameters
| Parameter | Type | Default | Description |
|---|---|---|---|
input |
string | required | Text to synthesize |
model |
string | server's model | Model to use (optional, should match server if specified) |
voice |
string | "vivian" | Speaker name (e.g., vivian, ryan, aiden) |
response_format |
string | "wav" | Audio format: wav, mp3, flac, pcm, aac, opus |
speed |
float | 1.0 | Playback speed (0.25-4.0, not supported with stream=true) |
vLLM-Omni Extension Parameters
| Parameter | Type | Default | Description |
|---|---|---|---|
task_type |
string | "CustomVoice" | Task: CustomVoice, VoiceDesign, or Base |
language |
string | "Auto" | Language (see supported languages below) |
instructions |
string | "" | Voice style/emotion instructions |
max_new_tokens |
int | 2048 | Maximum tokens to generate |
initial_codec_chunk_frames |
int | null | Per-request initial chunk size override for TTFA tuning. When null, IC is computed dynamically based on server load. |
stream |
bool | false | Stream raw PCM chunks as they are decoded (requires response_format="pcm") |
Supported languages: Auto, Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian
Voice Clone Parameters (Base task)
| Parameter | Type | Required | Description |
|---|---|---|---|
ref_audio |
string | Yes | Reference audio (HTTP URL, base64 data URL, or file:// URI with --allowed-local-media-path) |
ref_text |
string | No | Transcript of reference audio (for ICL mode) |
x_vector_only_mode |
bool | No | Use speaker embedding only (no ICL) |
Streaming
Set stream=true with response_format="pcm" to receive raw PCM audio chunks as they are decoded
(one chunk per Code2Wav window, default 25 frames; configurable in the stage config):
curl -X POST http://localhost:8091/v1/audio/speech \
-H "Content-Type: application/json" \
-d '{
"input": "Hello, how are you?",
"voice": "vivian",
"language": "English",
"stream": true,
"response_format": "pcm"
}' --no-buffer | play -t raw -r 24000 -e signed -b 16 -c 1 -
Constraints:
stream=truerequiresresponse_format="pcm"(raw 16-bit signed PCM, 24 kHz mono).speedadjustment is not supported when streaming.- Requires the server stage config to have
async_chunk: true(default inqwen3_tts.yaml).
Streaming Text Input (WebSocket)
The /v1/audio/speech/stream WebSocket endpoint accepts text incrementally, buffers and splits it at sentence boundaries, and generates audio per sentence.
When stream_audio=true, each sentence is emitted as audio.start, one or more binary PCM frames, and audio.done.
Quick Start
python streaming_speech_client.py \
--text "Hello world. How are you? I am fine."
python streaming_speech_client.py \
--text "Hello world. How are you? I am fine." \
--simulate-stt --stt-delay 0.1
WebSocket Protocol
Client -> Server:
{"type": "session.config", "voice": "Vivian", "task_type": "CustomVoice", "language": "Auto", "split_granularity": "sentence", "stream_audio": true, "response_format": "pcm"}
{"type": "input.text", "text": "Hello, how are you? "}
{"type": "input.done"}
Server -> Client:
{"type": "audio.start", "sentence_index": 0, "sentence_text": "Hello, how are you?", "format": "pcm", "sample_rate": 24000}
// binary PCM frame(s)
{"type": "audio.done", "sentence_index": 0, "total_bytes": 96000, "error": false}
{"type": "session.done", "total_sentences": 1}
Limitations
- Single request: Batch processing is not yet optimized for online serving.
Troubleshooting
- TTS model did not produce audio output: Ensure you're using the correct model variant for your task type (CustomVoice task → CustomVoice model, etc.)
- Connection refused: Make sure the server is running on the correct port
- Out of memory: Use smaller model variant (
Qwen/Qwen3-TTS-12Hz-0.6B-CustomVoice) or reduce--gpu-memory-utilization - Unsupported speaker: Use
/v1/audio/voicesto list available voices for the loaded model - Voice clone fails: Ensure you're using the Base model variant for voice cloning
Example materials
??? abstract "gradio_demo.py"
py --8<-- "examples/online_serving/qwen3_tts/gradio_demo.py"
??? abstract "gradio_fastrtc_demo.py"
py --8<-- "examples/online_serving/qwen3_tts/gradio_fastrtc_demo.py"
??? abstract "openai_speech_client.py"
py --8<-- "examples/online_serving/qwen3_tts/openai_speech_client.py"
??? abstract "run_gradio_demo.sh"
sh --8<-- "examples/online_serving/qwen3_tts/run_gradio_demo.sh"
??? abstract "run_server.sh"
sh --8<-- "examples/online_serving/qwen3_tts/run_server.sh"
??? abstract "streaming_speech_client.py"
py --8<-- "examples/online_serving/qwen3_tts/streaming_speech_client.py"
??? abstract "tts_common.py"
py --8<-- "examples/online_serving/qwen3_tts/tts_common.py"