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Grenmango/whisper-medium-en-vi-accent
whisper-medium-en-vi-accent is a automatic speech recognition model from Grenmango. Use it when you need speech turned into text. The card lists the license as apache-2.0.
This model is a fine-tuned, standalone merged version of openai/whisper-medium.en specifically adapted for English speech spoken with a Vietnamese accent.
Downloads · 30 days
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From the Hugging Face model README
This model is a fine-tuned, standalone merged version of openai/whisper-medium.en specifically adapted for English speech spoken with a Vietnamese accent.
It was trained on native Vietnamese speakers from the L2-ARCTIC speech corpus using Parameter-Efficient Fine-Tuning (LoRA rank=32, alpha=64), with the learned adapters permanently merged into the base model weights.
Evaluated against the baseline whisper-medium.en and whisper-large-v3-turbo on clean held-out read speech (test split):
| Model | Test Split WER | Test Split CER |
|---|---|---|
This Model (whisper-medium-en-vi-accent) | 11.42% | 6.05% |
openai/whisper-medium.en (Zero-Shot) | 18.85% | 9.97% |
openai/whisper-large-v3-turbo (Zero-Shot) | 15.99% | 8.16% |
whisper-medium.en.openai/whisper-medium.en (769M parameters)q_proj, k_proj, v_proj, out_proj, fc1, fc2)WhisperForConditionalGeneration)This is a standalone model. You can load and use it directly with Hugging Face transformers without needing peft or any extra setup.
pipeline (Recommended)from transformers import pipeline
# Initialize the pipeline
transcriber = pipeline(
"automatic-speech-recognition",
model="Grenmango/whisper-medium-en-vi-accent",
chunk_length_s=30,
device="cuda", # or "cpu"
)
# Transcribe an audio file (automatically resampled to 16kHz)
result = transcriber("path/to/audio.wav")
print(result["text"])
import torch
import soundfile as sf
from transformers import WhisperProcessor, WhisperForConditionalGeneration
model_id = "Grenmango/whisper-medium-en-vi-accent"
processor = WhisperProcessor.from_pretrained(model_id)
model = WhisperForConditionalGeneration.from_pretrained(model_id, torch_dtype=torch.float16, device_map="auto")
# Load 16kHz audio
audio_data, sample_rate = sf.read("path/to/audio.wav")
if sample_rate != 16000:
import soxr
audio_data = soxr.resample(audio_data, sample_rate, 16000)
input_features = processor(audio_data, sampling_rate=16000, return_tensors="pt").input_features.to("cuda", torch.float16)
# Generate transcription
predicted_ids = model.generate(input_features, max_new_tokens=128)
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
print("Transcription:", transcription)
This model is part of a complete multi-accent & personalized English fine-tuning suite on Hugging Face: