safetensors whisper generated_from_trainer automatic-speech-recognition en wo dataset:bilalfaye/english-wolof-french-dataset base_model:openai/whisper-medium base_model:finetune:openai/whisper-medium license:apache-2.0 region:us
---
base_model: openai/whisper-medium
datasets:
- bilalfaye/english-wolof-french-dataset
language:
- en
- wo
license: apache-2.0
metrics:
- bleu
pipeline_tag: automatic-speech-recognition
tags:
- generated_from_trainer
model-index:
- name: whisper-medium-english-2-wolof
  results: []
---

<!-- This model card has been generated automatically according to the information the Trainer had access to. You
should probably proofread and complete it, then remove this comment. -->

# whisper-medium-english-2-wolof

This model is a fine-tuned version of [openai/whisper-medium](https://huggingface.co/openai/whisper-medium) on the [bilalfaye/english-wolof-french-dataset](https://huggingface.co/datasets/bilalfaye/english-wolof-french-dataset). The model is designed to translate English audio into Wolof text. Since the base Whisper model does not natively support Wolof, this fine-tuned version bridges that gap.
It achieves the following results on the evaluation set:

- Loss: 1.1668
- Bleu: 34.6061

## Model Description

The model is based on OpenAI's Whisper architecture, fine-tuned to recognize and translate English speech to Wolof. It leverages the "medium" variant, offering a balance between accuracy and computational efficiency.

## Intended Uses & Limitations

**Intended uses:**
- Automatic transcription and translation of English audio into Wolof text.
- Assisting researchers and language learners working with English audio content.

**Limitations:**
- May struggle with heavy accents or noisy environments.
- Performance may vary depending on speaker pronunciation and recording quality.

## Training and Evaluation Data

The model was fine-tuned on the [bilalfaye/english-wolof-french-dataset](https://huggingface.co/datasets/bilalfaye/english-wolof-french-dataset), which consists of English audio paired with Wolof translations.

## Training Procedure

### Training hyperparameters

The following hyperparameters were used during training:
- learning_rate: 1e-05
- train_batch_size: 32
- eval_batch_size: 16
- seed: 42
- optimizer: Adam with betas=(0.9,0.999) and epsilon=1e-08
- lr_scheduler_type: linear
- lr_scheduler_warmup_steps: 500
- training_steps: 20000
- mixed_precision_training: Native AMP

### Training results

| Training Loss | Epoch  | Step  | Validation Loss | Bleu    |
|:-------------:|:------:|:-----:|:---------------:|:-------:|
| 0.9771        | 0.8941 | 2000  | 0.9736          | 22.8506 |
| 0.6832        | 1.7881 | 4000  | 0.8379          | 30.0113 |
| 0.4568        | 2.6822 | 6000  | 0.8083          | 33.4759 |
| 0.2623        | 3.5762 | 8000  | 0.8506          | 33.4723 |
| 0.1608        | 4.4703 | 10000 | 0.9128          | 33.6342 |
| 0.0758        | 5.3643 | 12000 | 0.9808          | 33.7770 |
| 0.0315        | 6.2584 | 14000 | 1.0546          | 34.0842 |
| 0.0133        | 7.1524 | 16000 | 1.1085          | 34.2531 |
| 0.0057        | 8.0465 | 18000 | 1.1455          | 34.5325 |
| 0.0046        | 8.9405 | 20000 | 1.1668          | 34.6061 |


### Framework versions

- Transformers 4.41.2
- Pytorch 2.4.0+cu121
- Datasets 3.2.0
- Tokenizers 0.19.1

## Inference

### Using Python Code

```python
! pip install transformers datasets torch

import torch
from transformers import WhisperForConditionalGeneration, WhisperProcessor
from datasets import load_dataset

# Load model and processor
device = "cuda:0" if torch.cuda.is_available() else "cpu"
model = WhisperForConditionalGeneration.from_pretrained("bilalfaye/whisper-medium-english-2-wolof").to(device)
processor = WhisperProcessor.from_pretrained("bilalfaye/whisper-medium-english-2-wolof")

# Load dataset
streaming_dataset = load_dataset("bilalfaye/english-wolof-french-dataset", split="train", streaming=True)
iterator = iter(streaming_dataset)
sample = next(iterator)
sample = next(iterator)
sample = next(iterator)


# Preprocess audio
input_features = processor(sample["en_audio"]["audio"]["array"],
                           sampling_rate=sample["en_audio"]["audio"]["sampling_rate"],
                           return_tensors="pt").input_features.to(device)

# Generate transcription
predicted_ids = model.generate(input_features)
transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)

print("Correct sentence:", sample["en"])
print("Transcription:", transcription[0])
```

### Using Gradio Interface

```python
! pip install gradio

from transformers import pipeline
import gradio as gr
import numpy as np


# Load model pipeline
device = "cuda:0" if torch.cuda.is_available() else "cpu"
pipe = pipeline(task="automatic-speech-recognition", model="bilalfaye/whisper-medium-english-2-wolof", device=device)

# Function for transcription
def transcribe(audio):
    if audio is None:
        return "No audio provided. Please try again."

    if isinstance(audio, str):  
        waveform, sample_rate = torchaudio.load(audio)
    elif isinstance(audio, tuple):  # Case microphone (Gradio donne un tuple (fichier, sample_rate))
        waveform, sample_rate = torchaudio.load(audio[0]) 
    else:
        return "Invalid audio input format."
    
    if waveform.shape[0] > 1:
        mono_audio = waveform.mean(dim=0, keepdim=True)
    else:
        mono_audio = waveform

    target_sample_rate = 16000
    if sample_rate != target_sample_rate:
        resampler = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=target_sample_rate)
        mono_audio = resampler(mono_audio)
        sample_rate = target_sample_rate

    mono_audio = mono_audio.squeeze(0).numpy().astype(np.float32)

    result = pipe({"array": mono_audio, "sampling_rate": sample_rate})
    return result['text']


# Create Gradio interfaces
interface = gr.Interface(
    fn=transcribe,
    inputs=gr.Audio(sources=["upload", "microphone"], type="filepath"),  
    outputs="text",
    title="Whisper Medium English Translation",
    description="Record audio in English and translate it to Wolof using a fine-tuned Whisper medium model.",
    #live=True,
)


app = gr.TabbedInterface(
    [interface],
    ["Use Uploaded File or Microphone"]  
)

app.launch(debug=True, share=True)
```

**Author**
  - Bilal FAYE