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Selest/wav2vec2-bert-assyrian-urmi-asr
wav2vec2-bert-assyrian-urmi-asr is a automatic speech recognition model from Selest. Use it when you need speech turned into text. It is set up for transformers. The card lists the license as apache-2.0.
Fine-tuned Wav2Vec2BertForCTC on Neo-Aramaic (Assyrian / Syriac tradition) — Urmi (Urmia) ASR data. Hub language: aii (Assyrian Neo-Aramaic). Base checkpoint: facebook/w2v-bert-2.0. Input: log-mel features via Seamles…
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From the Hugging Face model README
Fine-tuned Wav2Vec2BertForCTC on Neo-Aramaic (Assyrian / Syriac tradition) — Urmi (Urmia) ASR data. Hub language: aii (Assyrian Neo-Aramaic). Base checkpoint: facebook/w2v-bert-2.0.
Input: log-mel features via SeamlessM4TFeatureExtractor (see preprocessor_config.json), sampling rate 16000 Hz.
Веса в репозитории для выгрузки: model.safetensors (копия из best_model).
import torch
from transformers import Wav2Vec2BertForCTC, Wav2Vec2BertProcessor
import soundfile as sf
model_id = "<YOUR_HF_USERNAME>/<YOUR_REPO_NAME>"
processor = Wav2Vec2BertProcessor.from_pretrained(model_id)
model = Wav2Vec2BertForCTC.from_pretrained(model_id)
audio, sr = sf.read("audio.wav", dtype="float32")
if sr != 16000:
raise ValueError(f"Expected 16 kHz, got {sr}")
inputs = processor(audio, sampling_rate=16000, return_tensors="pt")
with torch.inference_mode():
logits = model(**inputs).logits
pred_ids = torch.argmax(logits, dim=-1)
text = processor.batch_decode(pred_ids)[0]
print(text)
Augmentation included speed perturbation (details in experiment repo / paper).
Model quality depends on recording conditions and vocabulary coverage; evaluate on your domain before production use.