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README.md
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```markdown
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---
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language: en
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license: mit
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model-index:
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- name: whisper-small-tr
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results:
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- task:
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type: automatic-speech-recognition
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name: Automatic Speech Recognition
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metrics:
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- type: wer
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value: 7.75
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name: Word Error Rate
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- type: cer
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value: 1.95
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name: Character Error Rate
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widget:
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- audio: https://huggingface.co/datasets/NgoHoang/Vietnamese_Speech_Recognition/resolve/main/Test/audio/common_voice_vi_24070014.mp3
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---
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# whisper-small-tr - Fine-tuned Whisper Small for Turkish ASR
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This model is a fine-tuned version of the `openai/whisper-small` base model by OpenAI, optimized for Turkish Automatic Speech Recognition (ASR).
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## Model Description
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Whisper models are powerful multilingual and multitask models pre-trained on a large variety of audio data. This project aims to significantly enhance the performance of the `whisper-small` model specifically for Turkish, by fine-tuning it on the `Codyfederer/tr-full-dataset` dataset.
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## Training Data
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The model was primarily trained on the Turkish audio and transcription dataset named `Codyfederer/tr-full-dataset`. From this dataset, 3000 samples were selected and split into 90% for training and 10% for testing.
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## Training Parameters
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The training was performed using the Hugging Face `Trainer` class with the following `Seq2SeqTrainingArguments`:
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- `output_dir`: `./whisper-small-tr`
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- `per_device_train_batch_size`: 16
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- `gradient_accumulation_steps`: 1
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- `learning_rate`: 3e-5
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- `warmup_steps`: 50
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- `num_train_epochs`: 3
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- `weight_decay`: 0.005
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- `gradient_checkpointing`: `True` (For memory optimization)
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- `fp16`: `True` (For faster training)
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- `eval_strategy`: `"steps"`
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- `per_device_eval_batch_size`: 8
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- `predict_with_generate`: `True`
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- `generation_max_length`: 225
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- `save_steps`: 200
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- `eval_steps`: 200
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- `logging_steps`: 25
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- `report_to`: `["tensorboard"]`
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- `load_best_model_at_end`: `True`
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- `metric_for_best_model`: `"wer"` (Lower is better)
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- `greater_is_better`: `False`
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- `push_to_hub`: `True`
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- `hub_model_id`: `whisper-small-tr`
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- `optim`: `adamw_torch`
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- `dataloader_num_workers`: 4
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- `dataloader_pin_memory`: `True`
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- `save_total_limit`: 2
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## Performance
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Evaluation results of the model on the test set:
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- **Word Error Rate (WER)**: 7.75%
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- **Character Error Rate (CER)**: 1.95%
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- **Loss**: 0.1321
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#### Comparison with Base Model (on example audio)
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In a comparison conducted with a new audio file (`/content/audio.mp3`):
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- **Base Whisper Model**: WER: 23.53% | CER: 2.82%
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- **Fine-Tuned Model**: WER: 11.76% | CER: 2.11%
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These results demonstrate a significant improvement in the fine-tuned model's performance for the Turkish ASR task compared to the base model.
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## How to Use
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You can easily use this model with the Hugging Face `transformers` library:
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```python
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from transformers import pipeline
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import torch
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# Load the model
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pipeline = pipeline(
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task="automatic-speech-recognition",
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model="emredeveloper/whisper-small-tr", # Your username/repo name
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chunk_length_s=30,
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device="cuda" if torch.cuda.is_available() else "cpu",
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)
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# Transcribe an audio file
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audio_file = "path/to/your/audio.flac" # Specify the path to your audio file
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text = pipeline(audio_file)["text"]
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print(text)
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```
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### Gradio Demo
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You can also create a Gradio demo to interactively test the model:
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```python
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import gradio as gr
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from transformers import pipeline
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import torch
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pipeline = pipeline(
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task="automatic-speech-recognition",
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model="emredeveloper/whisper-small-tr", # Your username/repo name
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chunk_length_s=30,
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device="cuda" if torch.cuda.is_available() else "cpu",
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)
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def transcribe(audio):
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if audio is None:
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return ""
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text = pipeline(audio)["text"]
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return text
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iface = gr.Interface(
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fn=transcribe,
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inputs=gr.Audio(sources=["microphone", "upload"], type="filepath"),
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outputs="text",
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title="Fine-Tuned Whisper Turkish Demo",
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description="Record your voice or upload a Turkish audio file to see the model in action.",
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)
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iface.launch()
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```
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