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from huggingface_hub import snapshot_download
from faster_whisper import WhisperModel
model_dir = snapshot_download("stcoats/whisper-large-v3-DASS2019-ct2")
model = WhisperModel(model_dir, device="cuda", compute_type="float16")
segments, info = model.transcribe("audio.wav", language="en", beam_size=5)
for s in segments:
print(f"[{s.start:.2f}-{s.end:.2f}] {s.text}")from huggingface_hub import snapshot_download
import whisperx
model_dir = snapshot_download("stcoats/whisper-large-v3-DASS2019-ct2")
model = whisperx.load_model(model_dir, device="cuda", compute_type="float16")
audio = whisperx.load_audio("audio.wav")
result = model.transcribe(audio, language="en", vad_filter=False)
for s in result["segments"]:
print(f"[{s['start']:.2f}-{s['end']:.2f}] {s['text']}")@inproceedings{coats-2026-fine,
title = {A Fine-tuned ASR Model for Historical American Dialect Recordings},
author = {Coats, Steven},
booktitle = {Proceedings of the Fifteenth Language Resources and Evaluation Conference (LREC 2026)},
month = {May},
year = {2026},
pages = {1372--1381},
address = {Palma, Mallorca, Spain},
publisher = {European Language Resources Association (ELRA)},
editor = {Piperidis, Stelios and Bel, Núria and van den Heuvel, Henk and Ide, Nancy and Krek, Simon and Toral, Antonio},
doi = {10.63317/5bjeqct6ozd3},
}