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weights-and-wires/audio8-tts-from-scratch
audio8-tts-from-scratch is a text-to-speech model from weights-and-wires. Use it when you need text read aloud. It is set up for transformers. The card lists the license as apache-2.0.
A DualAR text-to-speech language model trained from scratch on 440 clips (72.6 minutes) of LibriSpeech clean/train.100, 4 speaker(s).
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From the Hugging Face model README
A DualAR text-to-speech language model trained from scratch on
440 clips (72.6 minutes) of LibriSpeech
clean/train.100, 4 speaker(s).
⚠️ This is a training-pipeline experiment, not a production TTS model. The ~601 M-parameter transformer was randomly re-initialized and trained for 280 optimizer steps. Expect undertrained, partly unintelligible speech. LibriSpeech is 16 kHz upsampled to the codec's 44.1 kHz, which caps quality further.
| Component | Status |
|---|---|
| Slow AR transformer (24 layers, dim 896) | trained from random init |
| Fast AR transformer (4 layers) | trained from random init |
Neural codec (codec.pth, 10x4096 @ 44100 Hz) | reused frozen from Audio8/Audio8-TTS-Preview-0.6b |
| Tokenizer / architecture config | reused from the same checkpoint |
| Setting | Value |
|---|---|
| epochs | 10 |
| effective batch size | 16 |
| learning rate | 0.0002 (cosine, 0.05 warmup) |
| weight decay / grad clip | 0.01 / 1.0 |
| precision | fp32 weights + bf16 autocast |
| max sequence length | 2048 |
| train / eval examples | 440 / 23 |
| hardware | 1x NVIDIA A100-SXM4-40GB |
| Metric | Value |
|---|---|
| final train slow_loss | 3.9706 |
| final train fast_loss | 5.4779 |
| final train slow_accuracy | 0.4759 |
| final train fast_accuracy | 0.1876 |
| final eval slow_loss | 7.6749 |
| final eval fast_loss | 6.9021 |
Random-guess baselines for reference: slow_loss ln(vocab) = 11.96,
fast_loss ln(4096) = 8.32.
import soundfile as sf, torch
from transformers import AutoModel, AutoProcessor
repo = "weights-and-wires/audio8-tts-from-scratch"
processor = AutoProcessor.from_pretrained(repo, trust_remote_code=True)
model = AutoModel.from_pretrained(repo, trust_remote_code=True, dtype=torch.bfloat16).eval().cuda()
inputs = processor(text=["Hello world."], return_tensors="pt")
inputs = {k: v.cuda() for k, v in inputs.items()}
out = model.generate(**inputs, max_new_tokens=512, temperature=0.8, top_p=0.95,
top_k=50, do_sample=True, return_dict_in_generate=True)
audio, lengths = model.decode_audio(out.codes)
sf.write("out.wav", audio[0, :int(lengths[0])].float().cpu().numpy(), 44100)
The model was trained on the no-reference prompt template only — pass text= alone;
voice cloning via reference_audio= was never trained and will not work.
Apache-2.0, inherited from the base checkpoint. LibriSpeech is CC-BY-4.0.