# TinyWave Base Speech 2B

**TinyWave Base Speech 2B** is a compact speech-to-speech generation model distilled from the 7B SPIRIT-LM-Base teacher. It uses HuBERT-based phonetic tokens for efficient, high-quality speech generation and is optimized for **fast inference** on **commodity hardware**.

This model focuses on generating semantically coherent speech continuations without expressive modulation (e.g., pitch/style tokens). It is ideal for **low-resource speech agents**, **instruction-following speech bots**, and **embedded systems**.

> 📖 See the [TinyWave paper (arXiv:2506.23670)](https://arxiv.org/abs/2506.23670) and [demo site](https://mohammadmahdinoori.github.io/tinywave-landing/) for more details.

---

## 🔧 Usage

This model requires **SPIRIT-LM's base speech tokenizer**, which uses HuBERT units without pitch/style tokens.

### 1. Clone SPIRIT-LM and Install Requirements

```bash
git clone https://github.com/facebookresearch/spiritlm
cd spiritlm
pip install -e '.[eval]'
````

---

### 2. Load Tokenizer

```python
from spiritlm.speech_tokenizer import spiritlm_base
speech_tokenizer = spiritlm_base()
```

---

### 3. Inference Code (Speech-to-Speech)

```python
from transformers import LlamaForCausalLM, AutoTokenizer
import torchaudio
import torch

# Load model and tokenizer
MODEL_PATH = "tinywave/speech-base-2b"
tokenizer = AutoTokenizer.from_pretrained(MODEL_PATH)
model = LlamaForCausalLM.from_pretrained(MODEL_PATH, device_map="auto", torch_dtype=torch.bfloat16)

# Load base speech tokenizer
speech_tokenizer = spiritlm_base()

def get_inference(audio_path):
    audio, _ = torchaudio.load(audio_path)
    input_values = audio.view(1, 1, -1).to(speech_tokenizer.hubert_model.device).float()
    tokens = speech_tokenizer.encode_string(input_values)
    input_ids = tokenizer(tokens, return_tensors="pt").input_ids.to(model.device)
    output = model.generate(input_ids, max_new_tokens=256, top_p=0.9, temperature=0.9, do_sample=True)
    return tokenizer.decode(output[0])
```

---

### 4. Decode to WAV

```python
import numpy as np
from scipy.io.wavfile import write

def save_array_to_wav_int16(audio_array: np.ndarray, sampling_rate=16000, filename="output.wav"):
    scaled = np.int16(audio_array / np.max(np.abs(audio_array)) * 32767)
    write(filename, sampling_rate, scaled)

decoded_audio = speech_tokenizer.decode(generated_output.replace(" ", "").replace("<s>", "").replace("</s>", ""), speaker_id=2)
save_array_to_wav_int16(decoded_audio, filename="generated.wav")
```

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## 🗣️ Inference Example

### 🎧 Basic Speech Continuation

Input: `simple_prompt.wav`
Output: Semantically consistent speech continuation without expressive variation.

---

## 🧠 Model Details

| Feature             | Description                                      |
| ------------------- | ------------------------------------------------ |
| Architecture        | 2B parameter distilled transformer               |
| Tokenizer           | SPIRIT-LM Base (HuBERT phonetic tokens)          |
| Input Type          | Discrete HuBERT tokens only (speech-only)        |
| Output Type         | Discrete audio tokens                            |
| Teacher Model       | SPIRIT-LM-Base 7B                                |
| Tasks               | Speech continuation                              |
| Distillation Method | Layer-aligned (hidden states, attention, logits) |

---

## 📎 Citation

```bibtex
@article{nouriborji2025tinywave,
  title={Efficient Interleaved Speech Modeling through Knowledge Distillation},
  author={Nouriborji, Mohammadmahdi and Rohanian, Morteza},
  journal={arXiv preprint arXiv:2506.23670},
  year={2025}
}
```

---

## 📂 Resources

* 🔗 [Project Page](https://mohammadmahdinoori.github.io/tinywave-landing/)
* 💬 [Demo Samples](https://mohammadmahdinoori.github.io/tinywave-landing/#samples)
* 🧠 [Training & Codebase](https://github.com/mohammadmahdinoori/TinyWave)