babelvox 0.1.0

Real-time text-to-speech on Intel NPU/CPU via OpenVINO

BabelVox

Real-time text-to-speech on Intel NPU via OpenVINO. Runs Qwen3-TTS 0.6B inference entirely on Intel NPU (AI Boost), achieving RTF=1.0x (real-time) speech synthesis on a Lunar Lake ultrabook.

No PyTorch at runtime. Dependencies: openvino, numpy, librosa, soundfile, scipy, transformers (tokenizer only).

Installation

pip install babelvox

Or from source:

git clone https://github.com/YOUR_USERNAME/babelvox.git
cd babelvox
pip install -e .

Quick start

As a library

from babelvox import BabelVox

tts = BabelVox(
    export_dir="./openvino_export",
    device="NPU",
    precision="int8",
    use_cp_kv_cache=True,
    talker_buckets=[64, 128, 256],
)

wav, sr = tts.generate("Don't panic.", language="English")

import soundfile as sf
sf.write("output.wav", wav, sr)

From the command line

# Real-time on NPU with all optimizations
babelvox \
  --device NPU \
  --int8 \
  --cp-kv-cache \
  --talker-buckets "64,128,256" \
  --max-tokens 200 \
  --text "Hello, this is real-time speech synthesis on an Intel NPU." \
  --output hello.wav

# CPU-only (no NPU required, ~1.1x RTF)
babelvox \
  --device CPU \
  --int8 \
  --cp-kv-cache \
  --max-tokens 200 \
  --text "Hello world" \
  --output hello.wav

Model export (one-time setup)

BabelVox needs pre-exported OpenVINO IR models. The export scripts in tools/ require PyTorch and the original Qwen3-TTS model (~2.4 GB download):

pip install torch qwen-tts nncf

# Export OpenVINO IR models
python tools/export_tts_lm.py
python tools/export_speaker_encoder.py
python tools/export_decoder.py
python tools/export_tokenizer_encoder.py
python tools/export_cp_kvcache.py
python tools/export_weights.py

# Quantize to INT8 (recommended)
python tools/quantize_models.py --int8

After export, PyTorch is no longer needed.

Performance

Optimization progression

Optimization	RTF	Per-step	Notes
FP16 NPU baseline	3.0x	246 ms	Full-recompute, padded to 256 tokens
+ INT8 quantization	2.1x	156 ms	NNCF INT8_SYM weight compression
+ CP KV cache	1.4x	106 ms	Eliminates redundant code predictor recomputation
+ Multi-bucket talker	1.0x	~80 ms	Dynamically picks smallest NPU shape per step

RTF = Real-Time Factor. RTF=1.0x means generating 1 second of audio takes 1 second of compute.

Where the time goes (INT8 + CP KV cache, 256-token bucket)

Component	Device	Time	Share
Talker (28-layer transformer)	NPU	61 ms	57%
Code predictor (15 groups)	CPU	45 ms	43%
Numpy overhead (embeddings, sampling)	CPU	<1 ms	<1%

Multi-bucket scaling

The talker scales linearly with sequence length on NPU. Pre-compiling at multiple sizes and routing to the smallest bucket that fits dramatically reduces wasted compute:

Bucket size	Talker time	Total (+ 45ms CP)	Effective RTF
64	15 ms	60 ms	0.72x
128	22 ms	67 ms	0.80x
192	31 ms	76 ms	0.91x
256	43 ms	88 ms	1.06x

Hardware tested

• CPU: Intel Core Ultra 7 258V (Lunar Lake)
• NPU: Intel AI Boost (~13 TOPS)
• RAM: 32 GB LPDDR5x
• Device: Samsung Galaxy Book5 Pro

Architecture

Qwen3-TTS uses 5 model components orchestrated in an autoregressive loop:

Text --> Tokenizer --> Text Embeddings --> Talker (28L transformer) --> Codec code_0
                                               |
                       Speaker Embedding ------+    code_0 --> Code Predictor (5L) --> codes 1-15
                       (from reference audio)            \--> repeat 15x with KV cache
                                                                     |
                                           All 16 codes --> Tokenizer Decoder --> Waveform

Component	Layers	Hidden	Heads	Device	INT8 size
Talker	28	1024	16Q/8KV	NPU	444 MB
Code predictor	5	1024	16Q/8KV	CPU	79 MB
Tokenizer encoder	--	--	--	NPU	48 MB
Tokenizer decoder	--	--	--	NPU	114 MB
Speaker encoder	--	--	--	NPU	9 MB

CLI reference

Flag	Default	Description
--device	CPU	CPU or NPU
--int8	off	Use INT8 quantized models
--precision	fp16	fp16, int8, int4, or fp32
--cp-kv-cache	off	KV cache for code predictor (recommended)
--talker-buckets	none	Comma-separated NPU bucket sizes (e.g. 64,128,256)
--kv-cache	off	KV cache for talker (not recommended on NPU)
--max-tokens	200	Maximum generation steps
--max-talker-seq	256	Fixed talker padding (when not using buckets)
--max-decoder-frames	256	Max codec frames for audio decoder
--max-kv-len	256	KV cache buffer size (if --kv-cache)
--text	demo text	Text to synthesize
--language	English	Language for synthesis
--ref-audio	none	Reference audio for voice cloning
--output / -o	output.wav	Output WAV file path
--export-dir	openvino_export	Directory with exported models
--model-path	Qwen/Qwen3-TTS-12Hz-0.6B-Base	HuggingFace model (tokenizer)

Acknowledgments

Based on Qwen3-TTS by Alibaba Qwen Team (Apache-2.0).

License

Apache-2.0