| # TinyWave Interleaved Expressive 2B |
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| **TinyWave Interleaved Expressive 2B** is a compact, expressive speech-to-speech and speech-text language model distilled from the 7B SPIRIT-LM teacher. It supports **interleaved audio and text inputs** and is trained on 50k hours of public data using a multi-level layer-aligned distillation framework. |
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| Despite being 3Γ smaller than its teacher, the model retains **93β97%** of its accuracy on expressive benchmarks like StoryCloze and SALMon, and outperforms size-matched baselines. This model is ideal for **real-time multimodal agents**, **spoken dialogue systems**, and **low-resource deployment**. |
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| > π For more information, see the [TinyWave paper (arXiv:2506.23670)](https://arxiv.org/abs/2506.23670) and [project website](https://mohammadmahdinoori.github.io/tinywave-landing/). |
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| ## π§ Usage |
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| This model accepts interleaved speech and text inputs. It expects inputs to be encoded using SPIRIT-LMβs **expressive speech tokenizer**. |
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| ### 1. Clone SPIRIT-LM and Install Requirements |
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| ```bash |
| git clone https://github.com/facebookresearch/spiritlm |
| cd spiritlm |
| pip install -e '.[eval]' |
| ```` |
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| --- |
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| ### 2. Load Tokenizer |
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| ```python |
| from spiritlm.speech_tokenizer import spiritlm_expressive |
| speech_tokenizer = spiritlm_expressive() |
| ``` |
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| --- |
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| ### 3. Inference Code |
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| ```python |
| from transformers import LlamaForCausalLM, AutoTokenizer |
| import torchaudio |
| import torch |
| |
| # Load model and tokenizer |
| MODEL_PATH = "tinywave/interleaved-expressive-2b" |
| tokenizer = AutoTokenizer.from_pretrained(MODEL_PATH) |
| model = LlamaForCausalLM.from_pretrained(MODEL_PATH, device_map="auto", torch_dtype=torch.bfloat16) |
| |
| # Audio + Speech tokenizer |
| speech_tokenizer = spiritlm_expressive() |
| |
| def get_inference(input_audio_path): |
| audio, _ = torchaudio.load(input_audio_path) |
| input_values = audio.view(1, 1, -1).to(speech_tokenizer.hubert_model.device).float() |
| string_tokens = speech_tokenizer.encode_string(input_values) |
| input_ids = tokenizer(string_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]) |
| |
| # Text-based prompt |
| def get_inference_text(prompt): |
| input_ids = tokenizer(prompt + " [Speech]", 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]) |
| ``` |
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| --- |
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| ### 4. Decoding to WAV (optional) |
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| ```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(output_text.replace(" ", "").replace("<s>", "").replace("</s>", ""), speaker_id=2) |
| save_array_to_wav_int16(decoded_audio, filename="generated.wav") |
| ``` |
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| --- |
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| ## π£οΈ Inference Examples |
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| ### π§ Speech Continuation |
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| Input: `speech.wav` (spoken sentence) |
| Output: Expressive speech continuation in the same style and tone. |
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| --- |
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| ### π¬ Mixed Input: Text β Speech |
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| Prompt: |
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| ``` |
| "Once upon a time in a small village, a mysterious sound echoed through the forest. [Speech]" |
| ``` |
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| Output: Expressive spoken continuation in WAV format. |
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| --- |
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| ## π§ Model Details |
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| | Feature | Description | |
| | ------------------- | ------------------------------------------------- | |
| | Architecture | 2B parameter distilled transformer | |
| | Tokenizer | SPIRIT-LM Expressive (HuBERT + pitch/style) | |
| | Tasks | Speech continuation, mixed speech-text generation | |
| | Teacher Model | SPIRIT-LM-Expressive 7B | |
| | Distillation Method | Layer-aligned: hidden states, attention, logits | |
| | Input Types | Discrete HuBERT tokens and text | |
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| --- |
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| ## π Citation |
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| ```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} |
| } |
| ``` |
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| --- |
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| ## π Resources |
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| * π [Project Page](https://mohammadmahdinoori.github.io/tinywave-landing/) |
| * π¬ [Demo Samples](https://mohammadmahdinoori.github.io/tinywave-landing/#samples) |
| * π§ [Training & Codebase](https://github.com/mohammadmahdinoori/TinyWave) |