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Marwan-Kasem/whisper-medium-hi32
whisper-medium-hi32 is a automatic speech recognition model from Marwan-Kasem. Use it when you need speech turned into text. It is set up for adapter-transformers. The card lists the license as mit.
This repository hosts Whisper-Medium-hi32, a fine-tuned version of OpenAI's Whisper-Medium, developed by Marwan Kasem for automatic speech recognition (ASR) on clean, conversational English across diverse global accents.
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From the Hugging Face model README
This repository hosts Whisper-Medium-hi32, a fine-tuned version of OpenAI's Whisper-Medium, developed by Marwan Kasem for automatic speech recognition (ASR) on clean, conversational English across diverse global accents.
Whisper-Medium-hi32 is optimized for:
This model is trained to better handle the variability of everyday speech, making it ideal for call center analytics, transcription services, and research in sociolinguistics or dialectal speech.
| Metric | Value |
|---|---|
| Loss | 0.1971 |
| WER (Word Error Rate) | 16.52% |
These results were obtained on a clean, diverse English evaluation set featuring a wide range of accents.
openai/whisper-mediumThis ASR model is ideal for:
| Hyperparameter | Value |
|---|---|
learning_rate | 5e-4 |
train_batch_size | 2 |
eval_batch_size | 2 |
gradient_accumulation_steps | 4 |
total_train_batch_size | 8 |
optimizer | AdamW |
lr_scheduler | Linear |
warmup_steps | 500 |
mixed_precision | Native AMP |
seed | 42 |
| Epoch | Step | Train Loss | Val Loss | WER |
|---|---|---|---|---|
| 0.20 | 1250 | 0.3184 | 0.2267 | 19.12% |
| 0.40 | 2500 | 0.1957 | 0.2095 | 32.51% |
| 0.60 | 3750 | 0.1969 | 0.1971 | 16.52% ✅ |
v0.14.0v4.47.0v2.5.1+cu121v3.3.1v0.21.0from transformers import WhisperProcessor, WhisperForConditionalGeneration
import torchaudio
processor = WhisperProcessor.from_pretrained("MarwanKasem/whisper-medium-hi32")
model = WhisperForConditionalGeneration.from_pretrained("MarwanKasem/whisper-medium-hi32")
# Load and preprocess audio
speech_array, sampling_rate = torchaudio.load("sample.wav")
inputs = processor(speech_array[0], sampling_rate=sampling_rate, return_tensors="pt")
# Run inference
generated_ids = model.generate(inputs.input_features)
transcription = processor.batch_decode(generated_ids, skip_special_tokens=True)
print(transcription[0])