nano-qwen3tts-vllm

Qwen3-TTS with nano vLLM-style optimizations for fast text-to-speech generation.

Author note

This project was born after I tested the Qwen3-TTS model and realized the RTF (Real-Time Factor) remained stubbornly high, even on an H100. It sparked an idea: could we create a "nano-vllm" moment for Qwen3TTS? My goal is to match the efficiency of vLLM implementation within a concise, ~1k line codebase.

I am building this while diving deep into both the nano-vllm and Qwen3TTS source code. I anticipate some hurdles along the way and would appreciate any insights or feedback from the community.

Optimization for the Qwen3-TTS model will continue in the vllm-omni repo—stay tuned for updates!

Highlights

Important

🎉 Breakthrough: 8 concurrent audio streams can now run at near real-time speed on H100 for 1.7B model with new refactor now, enabling true parallel multi-user TTS service!

Performance Optimizations

• Continuous Batching — Batches multiple sequences and schedules prefill/decode across them for higher throughput.
• Page Attention — Paged KV cache with block tables and slot mapping for efficient memory use and variable-length sequences.
• CUDA Graph — Predictor and speech decoder use captured CUDA graphs (multiple batch sizes / decode lengths) to reduce kernel launch overhead.
• Streaming Support — Async generation with ZMQ: stream codec chunks as they are produced; API returns PCM audio stream (e.g. POST /v1/audio/speech with StreamingResponse).

Architecture (multiprocess + ZMQ)

When using multiprocess engines (USE_MULTIPROCESS_ENGINES=1 or the default server setup), the main process runs the API and orchestration; the talker and predictor models run in separate worker processes. Communication is over ZeroMQ (ZMQ) TCP sockets: main PUSHes commands and PULLs results; a result-bridge thread turns worker replies into asyncio Future completions so engine loops can dispatch to per-request queues.

flowchart TB
    subgraph Main["Main process"]
        API[FastAPI / StreamingResponse]
        IF[Interface & request_queues]
        TL[talker_loop_mp]
        PL[predictor_loop_mp]
        TC[TalkerWorkerClient\nPUSH commands]
        PC[PredictorWorkerClient\nPUSH commands]
        Bridge[Result bridge thread\nPULL results → Futures]
        API --> IF
        IF --> TL
        IF --> PL
        TL --> TC
        TL --> Bridge
        PL --> PC
        PL --> Bridge
        Bridge --> TL
        Bridge --> PL
    end

    subgraph Talker["Talker worker process"]
        TPULL[PULL\ncommands]
        TLLM[TalkerLLM]
        TPUSH[PUSH\nresults]
        TPULL --> TLLM --> TPUSH
    end

    subgraph Predictor["Predictor worker process"]
        PPULL[PULL\ncommands]
        PLLM[PredictorLLM]
        PPUSH[PUSH\nresults]
        PPULL --> PLLM --> PPUSH
    end

    TC -->|"add_request, run_step,\nclear_request (pickle)"| TPULL
    TPUSH -->|"step_id, outputs_all\n(pickle)"| Bridge
    PC -->|"commands (pickle)"| PPULL
    PPUSH -->|"results (pickle)"| Bridge

Loading

• Commands (main → workers): serialized with workers/protocol.py (pickle + numpy); e.g. add_request, run_step, clear_request, shutdown.
• Results (workers → main): worker PUSHes to main’s PULL sockets; the result-bridge thread completes the corresponding asyncio Future; engine loops then push (engine_type, msg_type, payload) into request_queues[request_id] for the API to consume.

Benchmark Results

Performance Comparison (Voice Design Model)

Tested on NVIDIA H100 with Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign with script examples/quick_benchmark.py:

Metric	nano-vllm	Original Qwen3-TTS	Improvement
Avg Generation Time	2.612s	8.487s	3.25x faster
Real-Time Factor (RTF)	0.399	1.467	3.68x better

Tested on NVIDIA L4 with Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign with script examples/quick_benchmark.py:

Metric	nano-vllm	Original Qwen3-TTS	Improvement
Avg Generation Time	4.319s	16.613s	3.85x faster
Real-Time Factor (RTF)	0.742	3.311	4.46x better

Key Findings:

• 🚀 4.86x faster generation speed
• 📊 RTF < 0.4 means nano-vllm generates audio 2.8x faster than real-time
• ⚡ Original implementation has RTF ~2.0 (slower than real-time)
• 💪 Consistent performance across different text lengths

Streaming Performance (Custom Voice Model)

L4 GPU, 0.6B custom model, decode wav each 1 chunk

Setup	First chunk latency (16 codec codes)	Inner chunk latency	RTF
1 CCU	160 ms	50 ms	0.65
2 CCUs	250 ms	90 ms	1.125

(CCU = concurrent request / “concurrent chunk unit” in the setup.)

Feature Completeness

• ✅ All Model Types Supported — CustomVoice (pre-defined speakers), VoiceDesign (text-to-voice), and Base (voice cloning)
• ✅ Voice Cloning — ICL mode and x_vector_only mode for reference audio-based voice cloning
• ✅ Voice Design — Generate voices from natural language descriptions
• ✅ Streaming Generation — Generator-based API for codec chunk streaming
• ✅ Multi-language — English, Chinese, and auto-detection support

Installation

Requirements

• Python ≥3.10
• PyTorch ≥2.10 with CUDA
• Compute capability ≥8.0 (e.g. Ampere/Ada) for Flash Attention
• qwen-tts, transformers, and other deps below

Flash Attention (recommended)

For a fast install without building from source, use pre-built wheels:

# Example: Python 3.12, CUDA 12.4, PyTorch 2.5
pip install https://github.com/mjun0812/flash-attention-prebuild-wheels/releases/download/v0.0.0/flash_attn-2.6.3+cu124torch2.5-cp312-cp312-linux_x86_64.whl

Pick the wheel that matches your Python, CUDA, and PyTorch from:
https://github.com/mjun0812/flash-attention-prebuild-wheels?target=https://github.com

Project

git clone https://github.com/tsdocode/nano-qwen3tts-vllm.git
cd nano-qwen3tts-vllm
uv sync
# or
pip install -e .

Supported Models

nano-qwen3tts-vllm supports all three Qwen3-TTS model types:

Model	Features	Language Support	Streaming	Instruction Control
Qwen3-TTS-12Hz-1.7B-VoiceDesign	Performs voice design based on user-provided descriptions.	Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian	✅	✅
Qwen3-TTS-12Hz-1.7B-CustomVoice	Provides style control over target timbres via user instructions; supports 9 premium timbres covering various combinations of gender, age, language, and dialect.	Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian	✅	✅
Qwen3-TTS-12Hz-1.7B-Base	Base model capable of 3-second rapid voice clone from user audio input; can be used for fine-tuning (FT) other models.	Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian	✅
Qwen3-TTS-12Hz-0.6B-CustomVoice	Supports 9 premium timbres covering various combinations of gender, age, language, and dialect.	Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian	✅
Qwen3-TTS-12Hz-0.6B-Base	Base model capable of 3-second rapid voice clone from user audio input; can be used for fine-tuning (FT) other models.	Chinese, English, Japanese, Korean, German, French, Russian, Portuguese, Spanish, Italian	✅

All models support both 12Hz (default, faster) and 25Hz (higher quality) variants.

💡 For complete, runnable examples, see the examples/ directory. Each example includes detailed usage instructions and demonstrates all features.

API Pattern

All generation methods return codec chunk generators for streaming support:

1. Generate codec chunks - Call generate_*() and wrap with list() to collect all chunks
2. Decode to audio - Use interface.speech_tokenizer.decode() to convert chunks to audio

This design enables:

• ✅ Streaming codec generation for low latency
• ✅ Consistent API across all model types
• ✅ Separation of generation and decoding steps
• ✅ Easy integration with streaming servers

Usage

1. Custom Voice (Pre-defined Speakers)

Generate speech with built-in speaker voices (e.g., Vivian, Mike, Sarah, etc.):

from nano_qwen3tts_vllm.interface import Qwen3TTSInterface
import soundfile as sf

# Load CustomVoice model
interface = Qwen3TTSInterface.from_pretrained(
    pretrained_model_name_or_path="Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice",
    enforce_eager=False,
    tensor_parallel_size=1,
)

# Generate codec chunks
audio_codes = list(interface.generate_custom_voice(
    text="Hello, this is a test.",
    language="English",
    speaker="Vivian",
))

# Decode to audio using interface's built-in speech tokenizer
wavs, sr = interface.speech_tokenizer.decode([{"audio_codes": audio_codes}])
sf.write("output.wav", wavs[0], sr)

Available speakers: Vivian, Mike, Sarah, Laura, Alex, Ethan, Emma, and more (see model card).

2. Voice Design (Text-to-Voice)

Create voices from text descriptions:

from nano_qwen3tts_vllm.interface import Qwen3TTSInterface
import soundfile as sf

# Load VoiceDesign model
interface = Qwen3TTSInterface.from_pretrained(
    pretrained_model_name_or_path="Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign",
    enforce_eager=False,
    tensor_parallel_size=1,
)

# Generate with voice design instruction
audio_codes = list(interface.generate_voice_design(
    text="Hi! How are you doing today?",
    language="English",
    instruct="A young woman with a warm, friendly voice and slight excitement",
))

# Decode to audio using interface's built-in speech tokenizer
wavs, sr = interface.speech_tokenizer.decode([{"audio_codes": audio_codes}])
sf.write("output_designed.wav", wavs[0], sr)

See: examples/voice_design_example.py for more examples.

3. Voice Clone (Reference Audio)

Clone voices from reference audio samples:

from nano_qwen3tts_vllm.interface import Qwen3TTSInterface
import soundfile as sf

# Load Base model
interface = Qwen3TTSInterface.from_pretrained(
    pretrained_model_name_or_path="Qwen/Qwen3-TTS-12Hz-1.7B-Base",
    enforce_eager=False,
    tensor_parallel_size=1,
)

# Load reference audio
ref_audio, ref_sr = sf.read("reference.wav")

# Create voice clone prompt (ICL mode - with reference text)
voice_clone_prompt = interface.create_voice_clone_prompt(
    ref_audio=(ref_audio, ref_sr),
    ref_text="This is the reference text that was spoken in the audio.",
    x_vector_only_mode=False,  # ICL mode for better quality
)

# Generate codec chunks with cloned voice
audio_codes = list(interface.generate_voice_clone(
    text="Hello, this is a cloned voice speaking.",
    language="English",
    voice_clone_prompt=voice_clone_prompt,
))

# Decode to audio using interface's built-in speech tokenizer
wavs, sr = interface.speech_tokenizer.decode([{"audio_codes": audio_codes}])
sf.write("output_cloned.wav", wavs[0], sr)

Voice Clone Modes:

• ICL mode (x_vector_only_mode=False): Uses both speaker embedding and reference audio codes. Requires ref_text. Better quality and more accurate voice matching.
• x_vector_only mode (x_vector_only_mode=True): Uses only speaker embedding. No ref_text needed. Faster but less accurate.

See: examples/voice_clone_example.py for comprehensive examples with ICL and x_vector modes.

Streaming (multiprocess)

The server uses multiprocess engines (talker + predictor in separate processes) and streams codec chunks; POST /v1/audio/speech returns a streaming PCM response.

# Client: stream PCM and write to file (see examples/client.py)
import requests
r = requests.post(
    "http://localhost:8000/v1/audio/speech",
    json={"text": "Hello world.", "language": "English", "speaker": "Vivian"},
    stream=True,
)
# consume r.iter_content() and write to WAV

Run server

export QWEN3_TTS_MODEL_PATH=/path/to/qwen3tts
python -m uvicorn examples.server:app --host 0.0.0.0 --port 8000
# or
python examples/server.py

Options

Parameter	Description
model_path	Path to Qwen3-TTS model (custom voice)
enforce_eager	Disable CUDA graphs (for debugging)
tensor_parallel_size	Number of GPUs (1–8)
(multiprocess)	Talker and predictor run in separate processes (server)
QWEN3_TTS_MODEL_PATH	Model directory (server env)

Examples

📁 See examples/ for complete, runnable code!

Comprehensive example scripts are provided in the examples/ directory:

• custom_voice_example.py - Pre-defined speaker voices
• voice_design_example.py - Voice design from text descriptions
• voice_clone_example.py - Voice cloning with ICL and x_vector modes
• server.py - FastAPI server with streaming support
• client.py - Client for streaming API

Running Examples

# Custom Voice
python examples/custom_voice_example.py \
    --model-path Qwen/Qwen3-TTS-12Hz-1.7B-CustomVoice \
    --text "Hello world" \
    --speaker Vivian

# Voice Design
python examples/voice_design_example.py \
    --model-path Qwen/Qwen3-TTS-12Hz-1.7B-VoiceDesign \
    --output-dir output

# Voice Clone
python examples/voice_clone_example.py \
    --model-path Qwen/Qwen3-TTS-12Hz-1.7B-Base \
    --ref-audio reference.wav \
    --ref-text "Reference transcript" \
    --output-dir output

Advanced Features

Using Local Models

If you have models downloaded locally:

interface = Qwen3TTSInterface(
    model_path="/path/to/local/model",
    enforce_eager=False,
    tensor_parallel_size=1,
)

Non-Streaming Mode

For better quality in non-real-time scenarios:

audio_codes = list(interface.generate_voice_design(
    text="Hello world",
    language="English",
    instruct="professional voice",
    non_streaming_mode=True,  # Better for offline processing
))
wavs, sr = interface.speech_tokenizer.decode([{"audio_codes": audio_codes}])

Supported Languages

All models support multiple languages:

• English - Full support
• Chinese (中文) - Full support with dialect variants
• Auto - Automatic language detection

Example with Chinese:

audio_codes = list(interface.generate_voice_design(
    text="你好，世界！",
    language="Chinese",
    instruct="温暖的女声",
))
wavs, sr = interface.speech_tokenizer.decode([{"audio_codes": audio_codes}])

Performance Tips

1. Use CUDA Graphs (enforce_eager=False) for 2-3x speedup
2. Use 12Hz models for faster generation (25Hz for higher quality)
3. Enable streaming mode (ZMQ) for lowest latency in production
4. Use generators — Process codec chunks as they're generated instead of collecting all at once

Further Optimization (contributions welcome)

• ✅ Support for all model types (CustomVoice, VoiceDesign, Base)
• ✅ Voice clone with ICL and x_vector modes
• ✅ Voice design from text descriptions
• ⏳ Make prefill stage run with CUDA Graph

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