---
license: mit
library_name: vibevoice.cpp
tags:
  - tts
  - asr
  - speech
  - vibevoice
  - gguf
  - ggml
base_model:
  - microsoft/VibeVoice-Realtime-0.5B
  - microsoft/VibeVoice-ASR
---

# vibevoice.cpp — quantized model bundle

**Brought to you by the [LocalAI](https://github.com/mudler/LocalAI) team** — the creators of LocalAI, the open-source AI engine that runs any model — LLMs, vision, voice, image, video — on any hardware. No GPU required.

Quantized GGUF weights for [vibevoice.cpp](https://github.com/mudler/vibevoice.cpp),
a C/C++ port of Microsoft VibeVoice (TTS + ASR) on top of `ggml`.

| File | Source | Quant | Size |
| ---- | ------ | ----- | ---- |
| `vibevoice-realtime-0.5B-q8_0.gguf` | `microsoft/VibeVoice-Realtime-0.5B` | Q8_0 (matmul) + F16 | ~1.6 GB |
| `vibevoice-asr-q8_0.gguf`           | `microsoft/VibeVoice-ASR`           | Q8_0 (matmul) + F16 | ~13 GB |
| `voice-en-Carter_man.gguf`          | upstream voice prompt cache         | F16                  | 8 MB |
| `voice-en-Emma.gguf`                | upstream voice prompt cache         | F16                  | 6 MB |
| `tokenizer.gguf`                    | Qwen2.5 BPE + VibeVoice specials    | —                    | 6 MB |

## Quantization scheme

`scripts/quantize_gguf.py` in the source repo selectively quantizes only the
LM matmul weights — attention q/k/v/o, ffn gate/up/down, and lm_head — to
Q8_0. Everything else (1-D conv kernels, RMSNorm scales, biases,
layer-scale gammas, token embeddings, small scalars) passes through
unchanged. The conv1d implementation in vibevoice.cpp casts kernels to F16
inline rather than dequantizing on the fly, so quantizing those would
corrupt the convolution outputs.

Q8_0 was chosen because it's pure-Python implementable in `gguf-py` and
gives a ~60% size reduction on the 7B ASR model with no measurable
quality regression in the closed-loop TTS → ASR roundtrip test.

## Quickstart

```bash
git clone --recursive https://github.com/mudler/vibevoice.cpp
cd vibevoice.cpp && cmake -B build -DVIBEVOICE_BUILD_TESTS=ON && cmake --build build -j

# Pull this bundle
mkdir -p models && cd models
hf download mudler/vibevoice.cpp-models --local-dir .
cd ..

# TTS
build/bin/vibevoice-cli tts \
    --model models/vibevoice-realtime-0.5B-q8_0.gguf \
    --voice models/voice-en-Carter_man.gguf \
    --tokenizer models/tokenizer.gguf \
    --text "Hello world this is a test of the synthesis system." \
    --out hello.wav

# ASR
build/bin/vibevoice-cli asr \
    --model models/vibevoice-asr-q8_0.gguf \
    --tokenizer models/tokenizer.gguf \
    --audio hello.wav
# -> [{"Start":0,"End":2.8,"Speaker":0,"Content":"Hello world, this is a test of the synthesis system."}]
```

## Closed-loop verification

The `test_closed_loop` ctest in vibevoice.cpp runs TTS → ASR end-to-end
and asserts ≥80% source-word recall in the recovered transcript. With
this bundle (both Q8_0 models) it passes at 10/10 (100 %).

## License

Weights are derived from Microsoft VibeVoice
([VibeVoice-Realtime-0.5B](https://huggingface.co/microsoft/VibeVoice-Realtime-0.5B)
and [VibeVoice-ASR](https://huggingface.co/microsoft/VibeVoice-ASR));
follow the upstream model licenses for use. The conversion + quantization
tooling is released under MIT as part of vibevoice.cpp.