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Audio8-TTS-0.1B (Preview)

A 170M-parameter zero-shot speech model designed for edge deployment

September 10, 2026

The smallest zero-shot TTS worth running.
A 170M-parameter zero-shot speech model designed for edge deployment.

0.1B targets local CPUs, low-memory devices, and offline environments. It compresses zero-shot voice conditioning into a 170M-parameter main generative model and provides a complete inference path around ONNX Runtime. Its value is retaining a usable voice-generation pipeline within a compact budget.

Keep DualAR, replace the Slow AR backbone

0.1B retains the same semantic-to-acoustic handoff as 0.6B. Slow AR advances semantic tokens frame by frame; Fast AR reads its hidden states and completes the acoustic codebooks across ten RVQ layers. The difference is that Slow AR uses the Falcon-H1 hybrid backbone, combining attention layers with Mamba 2 state-space layers and reducing width to 512 while preserving the contract between long-range state and within-frame acoustic decoding.

ComponentConfiguration
Main model170M parameters, excluding the audio decoder
Slow AR24 layers, width 512, 8 attention heads, 2 KV heads; Mamba d_state=64
Fast AR4 layers, width 512
Acoustic representation10 codebooks, 4,096 entries each
Neural audio codec44.1 kHz, 2,048 samples per model frame; about 120M parameters
ContextUp to 2,048 packed text/audio positions

170M describes the main generative model; the separate codec decoder still counts toward storage and memory when producing a waveform.

ONNX INT8: make the edge path real

Audio8-TTS-0.1B ONNX INT8 is more than quantized weights. It combines the Slow AR and Fast AR token-by-token graphs, an FP16 audio codec, explicit loop state, and a CPU inference interface into a path scheduled by ONNX Runtime. After downloading the model, CUDA, PyTorch, and Transformers are not required.

ModuleRuntime formRole
Slow ARINT8 token-by-token graphExplicitly passes attention and Mamba states; valid-prefix attention limits computation to the valid prefix
Fast ARINT8 token-by-token graphGenerates ten acoustic codebooks frame by frame
Audio encoder / decoderFP16Registers reference timbre and reconstructs 44.1 kHz waveforms
Output waveformFP32, monoPassed to a local file, PCM stream, or service interface
Edge runtime reference

On a Linux x86_64 test platform, ordinary inference after loading uses about 0.4 GiB. On an 8-thread host, Slow AR takes about 19 ms per token and Fast AR about 8 ms per frame; these figures vary with the ONNX Runtime build, allocator, thread count, and hardware.

Move voice registration out of the hot path

Ordinary synthesis needs only Slow AR, Fast AR, and the codec decoder. The codec encoder is loaded only when creating a new voice, converting a reference recording and accurate text into reusable voice codes; after registration it can be unloaded before normal synthesis resumes. Voice registration is a one-time lifecycle operation, not a resident cost for every request.

From model files to the service boundary, the CLI, local HTTP, streaming PCM, voice registration, and OpenAI-compatible interface are all covered.

Preview boundaries

0.1B primarily targets Chinese and English; other languages are experimental. INT8 quantization can change the sampled token sequence, so the ONNX version should be evaluated separately from the original PyTorch checkpoint. Zero-shot cloning still requires a clean reference recording, accurate text, and speaker authorization.