Step-Audio2 Offline Inference Examples
This directory contains examples for running offline inference with Step-Audio2 using vLLM-Omni.
Model Overview
Step-Audio2 is a two-stage audio model:
-
Stage 0 (Thinker): Audio understanding → Text + Audio tokens
- Input: Audio (16kHz)
- Output: Text transcription + Audio tokens for synthesis
-
Stage 1 (Token2Wav): Audio synthesis
- Input: Audio tokens + Speaker prompt wav
- Output: Synthesized audio waveform (24kHz)
Hardware Requirements
| Mode | GPU Configuration | VRAM Required |
|---|---|---|
| ASR (S2T) | 1x GPU | ~20-25GB |
| TTS/S2ST (single GPU) | 1x GPU | ~40-50GB |
| TTS/S2ST (multi GPU) | 2x GPU | GPU0: ~28GB, GPU1: ~22GB |
Tested on:
- 1x NVIDIA H100 80GB (single-card S2ST)
- 2x NVIDIA A10 40GB (multi-card S2ST)
Notes:
- Single GPU mode requires high VRAM due to both stages sharing memory
- Multi GPU mode separates Stage 0 (Thinker) and Stage 1 (Token2Wav) across GPUs
- VRAM usage can be adjusted via
gpu_memory_utilizationin stage config
Performance Benchmark
vLLM-Omni vs Official Step-Audio2
Single request latency comparison between vLLM-Omni and official Step-Audio2 implementation.
| Task | Tokens | vllm-omni | Step-Audio2 | Speedup |
|---|---|---|---|---|
| S2ST | ~85 | 5.45s | 7.36s | 1.35x |
| S2ST | ~160 | 6.67s | 13.92s | 2.09x |
| S2ST | ~315 | 9.42s | 31.50s | 3.34x |
| TTS | ~1024 | 16.65s | ~87s | ~5.2x |
Key observations:
- Speedup increases with sequence length due to vLLM's efficient KV cache management
- TTS (pure generation) shows the largest speedup (~5x)
- S2ST benefits from optimized multi-stage pipeline
Benchmark environment:
- GPU: NVIDIA H100 80GB (single card)
- Model: Step-Audio2-mini
- Warmup: 1 run, Measured: 3 runs (averaged)
Async Chunk Streaming Performance
Comparison between sequential (non-async) and async chunk modes via /v1/audio/speech TTS endpoint.
| Mode | Mean TTFP | Mean E2E | Mean RTF | Audio Throughput |
|---|---|---|---|---|
| Sequential | 4316ms | 4316ms | 0.938 | 1.07x realtime |
| Async Chunk | 1437ms | 4362ms | 0.949 | 1.06x realtime |
| Improvement | -67% (3x faster) | ~same | ~same | ~same |
Key observations:
- Async chunk reduces time-to-first-audio (TTFP) by 67% (4.3s → 1.4s)
- E2E latency remains comparable — async chunk overlaps Thinker decode with Token2Wav synthesis
- RTF < 1 in both modes (real-time capable)
- Sequential mode: TTFP ≈ E2E (must wait for all audio tokens before synthesis starts)
- Async chunk mode: Token2Wav starts after first 28 tokens (chunk_size=25 + lookahead=3)
Benchmark environment:
- GPU: 4x NVIDIA RTX 3090 24GB (TP=2 for Thinker, 1 GPU for Token2Wav)
- Model: Step-Audio2-mini
- Endpoint:
/v1/audio/speech(10 prompts, concurrency=1) - Measured via
bench_tts_serve.py
Installation
Make sure you have installed vLLM-Omni and all required dependencies:
# Install vLLM-Omni
pip install vllm-omni
# Install Step-Audio2 (REQUIRED for Token2Wav stage)
pip install step-audio2
Model Setup
Option 1: Auto-download from HuggingFace (Recommended)
The script will automatically download the model on first run:
# Just run without specifying --model, it will auto-download stepfun-ai/Step-Audio2-mini
python end2end.py --query-type audio_to_text
# Or explicitly specify the HuggingFace model
python end2end.py --query-type audio_to_text --model stepfun-ai/Step-Audio2-mini
Models will be cached in ~/.cache/huggingface/hub/ for future use.
Available models:
stepfun-ai/Step-Audio2-mini(smaller, faster)stepfun-ai/Step-Audio2-7B(larger, better quality)
Option 2: Manual Download (for offline use)
Download and use locally:
# Download from HuggingFace
huggingface-cli download stepfun-ai/Step-Audio2-mini --local-dir ./models/Step-Audio2-mini
# Then use the local path
python end2end.py --query-type audio_to_text --model ./models/Step-Audio2-mini
Ensure the model directory contains:
Step-Audio2-mini/
├── config.json
├── model.safetensors (or pytorch_model.bin)
├── tokenizer.json
├── tokenizer_config.json
└── token2wav/ # Token2Wav models (REQUIRED)
├── speech_tokenizer_v2_25hz.onnx # Audio tokenizer
├── campplus.onnx # Speaker encoder
├── flow.yaml # Flow model config
├── flow.pt # Flow model weights
└── hift.pt # HiFT vocoder weights
Usage Examples
1. Audio to Text (ASR - Speech Recognition)
Transcribe audio to text:
# Quick start - Using default model and test audio
python end2end.py --query-type audio_to_text
# Using your own audio file (model will auto-download)
python end2end.py --query-type audio_to_text \
--audio-path /path/to/input.wav
# With specific model
python end2end.py --query-type audio_to_text \
--audio-path /path/to/input.wav \
--model stepfun-ai/Step-Audio2-7B
# With custom question
python end2end.py --query-type audio_to_text \
--audio-path input.wav \
--question "What is the speaker saying?"
Output: Text transcription saved to output_step_audio2/00000_text.txt
2. Text to Audio (TTS - Speech Synthesis)
Convert text to speech:
# Basic TTS (model auto-downloads)
python end2end.py --query-type text_to_audio \
--text "Hello, this is a test of Step Audio 2 synthesis."
# With specific model
python end2end.py --query-type text_to_audio \
--text "Hello, this is a test." \
--model stepfun-ai/Step-Audio2-7B
Note: Speaker voice is controlled by the STEP_AUDIO2_DEFAULT_PROMPT_WAV environment variable or the default prompt wav bundled with the model.
Output:
- Text:
output_step_audio2/00000_text.txt - Audio:
output_step_audio2/00000_output.wav(24kHz)
3. Audio to Audio (Voice Conversion)
Process input audio and generate output audio:
# Basic voice conversion (model auto-downloads)
python end2end.py --query-type audio_to_audio \
--audio-path /path/to/source_audio.wav
# With specific model
python end2end.py --query-type audio_to_audio \
--audio-path source.wav \
--model stepfun-ai/Step-Audio2-7B
This mode:
- Understands the content in
--audio-path(source) - Generates audio output with the default voice
Note: To use a custom speaker voice, set the STEP_AUDIO2_DEFAULT_PROMPT_WAV environment variable.
Advanced Options
# Use custom stage configuration
python end2end.py --query-type audio_to_text \
--stage-configs-path /path/to/custom_config.yaml
# Multiple prompts (for batch testing)
python end2end.py --query-type audio_to_text \
--audio-path input.wav \
--num-prompts 5
# Custom output directory
python end2end.py --query-type text_to_audio \
--text "Test synthesis" \
--output-dir ./my_outputs
# Enable detailed logging
python end2end.py --query-type audio_to_text \
--audio-path input.wav \
--enable-stats
# Adjust generation parameters
python end2end.py --query-type audio_to_text \
--audio-path input.wav \
--max-tokens 2048
# Use custom speaker voice via environment variable
STEP_AUDIO2_DEFAULT_PROMPT_WAV=/path/to/speaker.wav python end2end.py \
--query-type text_to_audio \
--text "Hello world"
# Use Ray backend for distributed processing
python end2end.py --query-type text_to_audio \
--text "Hello world" \
--worker-backend ray \
--ray-address "auto"
Configuration
Stage Configuration
The default configuration (step_audio_2.yaml) uses:
- Stage 0 (Thinker): GPU 0, 80% memory
- Stage 1 (Token2Wav): GPU 1, 30% memory
For single GPU setup, edit the config to use devices: "0" for both stages.
Sampling Parameters
-
Thinker (Stage 0):
- Temperature: 0.7 (balanced creativity)
- Top-p: 0.9
- Max tokens: 1024 (configurable)
-
Token2Wav (Stage 1):
- Temperature: 0.0 (deterministic)
- Operates in generation mode (not sampling)
Common Issues
1. ImportError: No module named 's3tokenizer'
Solution: Install Step-Audio2 package:
pip install step-audio2
2. FileNotFoundError: prompt_wav file not found
Solution: Set the STEP_AUDIO2_DEFAULT_PROMPT_WAV environment variable to a valid audio file:
export STEP_AUDIO2_DEFAULT_PROMPT_WAV=/path/to/speaker.wav
python end2end.py --query-type text_to_audio --text "Hello"
Or ensure the default prompt wav (default_female.wav) exists in your model directory.
3. FileNotFoundError: token2wav models not found
Solution: Ensure your model directory has the complete token2wav/ subdirectory with all ONNX and PyTorch models.
4. CUDA Out of Memory
Solutions:
- Use single GPU mode (set both stages to
devices: "0") - Reduce
gpu_memory_utilizationin config - Reduce
max_num_batched_tokens - Process fewer prompts at once
5. Model not found in registry
Solution: Ensure you're using vLLM-Omni's entry point with --omni flag or install vllm-omni properly:
pip install vllm-omni
Output Files
The script generates files in the output directory (default: output_step_audio2/):
output_step_audio2/
├── 00000_text.txt # Text output from Thinker stage
├── 00000_output.wav # Audio output from Token2Wav stage (24kHz)
├── 00001_text.txt # (if multiple prompts)
└── 00001_output.wav
Performance Tips
- First run is slow: Stage initialization takes 20-60 seconds
- Single GPU: Set both stages to
devices: "0"in config - Multiple prompts: Use
--num-prompts Nfor batch testing - Ray backend: For multi-node or advanced scheduling
- Logging: Use
--enable-statsto debug performance issues
Speaker Voice Configuration
The Token2Wav stage requires a speaker prompt wav for voice conditioning. It is automatically resolved in this order:
STEP_AUDIO2_DEFAULT_PROMPT_WAVenvironment variable (if set){model_dir}/assets/default_female.wav{model_dir}/default_female.wav
If none are found, set the environment variable explicitly:
export STEP_AUDIO2_DEFAULT_PROMPT_WAV=/path/to/speaker.wav
Guidelines for custom speaker prompt:
- Duration: 3-10 seconds recommended
- Quality: Clean audio, minimal background noise
- Format: WAV, MP3, FLAC (will be resampled internally)
- Content: Clear speech, representative of target voice
Example Workflow
Complete example from audio to final output:
# 1. ASR: Transcribe audio
python end2end.py --query-type audio_to_text \
--audio-path interview.wav \
--model ./models/Step-Audio2-7B \
--output-dir ./outputs
# 2. Check the transcription
cat ./outputs/00000_text.txt
# 3. TTS: Synthesize new speech (with custom voice)
STEP_AUDIO2_DEFAULT_PROMPT_WAV=./speaker_samples/female_voice.wav \
python end2end.py --query-type text_to_audio \
--text "The quick brown fox jumps over the lazy dog" \
--model ./models/Step-Audio2-7B \
--output-dir ./outputs
# 4. Listen to the result
# Audio saved to: ./outputs/00000_output.wav