# Whisper to Oliver
**Fine-tuned Whisper for Real-World Conversational Audio**
[](https://huggingface.co/olib-ai/whisper-to-oliver)
[](https://opensource.org/licenses/MIT)
[](https://www.olib.ai)
## 🎯 Model Description
**Whisper to Oliver** is a specialized fine-tuned version of OpenAI's `whisper-large-v3-turbo` model, optimized for real-world conversational audio with challenging acoustic conditions. This model is specifically designed to excel at transcribing phone calls and conversations where audio quality may be compromised.
### ✨ Key Features
- 🎙️ **Enhanced Performance on Poor Quality Audio**: Fine-tuned on 170K conversational datasets with minor to poor audio quality
- 📞 **Phone Call Optimized**: Specifically trained on short conversational segments typical of phone calls
- 🚀 **Turbo Performance**: Inherits the speed advantages of whisper-large-v3-turbo
- 💼 **Enterprise Ready**: Developed by [Olib AI](https://www.olib.ai) for business applications
- 🔧 **FP32 Precision**: Full precision model for maximum accuracy
## 📊 Training Details
- **Base Model**: [openai/whisper-large-v3-turbo](https://huggingface.co/openai/whisper-large-v3-turbo)
- **Training Dataset**: 170,000 conversational audio samples
- **Audio Characteristics**: Minor to poor quality recordings
- **Focus**: Short conversational segments typical of phone interactions
- **Developer**: [Olib AI](https://www.olib.ai) - Building AI Services for Businesses
## 🚀 Usage
### Using the Transformers Library
```python
import torch
from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
device = "cuda:0" if torch.cuda.is_available() else "cpu"
torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
model_id = "olib-ai/whisper-to-oliver"
model = AutoModelForSpeechSeq2Seq.from_pretrained(
model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
)
model.to(device)
# Note: This model is in FP32 format
processor = AutoProcessor.from_pretrained(model_id)
pipe = pipeline(
"automatic-speech-recognition",
model=model,
tokenizer=processor.tokenizer,
feature_extractor=processor.feature_extractor,
torch_dtype=torch_dtype,
device=device,
)
# Transcribe audio
result = pipe("audio.mp3")
print(result["text"])
```
### Advanced Usage with Parameters
```python
# For better results with phone calls or poor quality audio
result = pipe(
"phone_call.mp3",
chunk_length_s=30,
batch_size=16,
return_timestamps=True,
)
print(result["text"])
```
## 📈 Performance
Whisper to Oliver shows significant improvements over the base model when dealing with:
- 📞 Phone call recordings
- 🎙️ Low-quality microphone inputs
- 🌐 Conversational speech with background noise
- 💬 Short dialogue segments
## 🎯 Intended Use
This model is designed for:
- Customer service call transcription
- Meeting transcription with variable audio quality
- Voice assistant applications
- Real-time conversation analysis
- Accessibility applications for hearing-impaired users
## ⚠️ Limitations and Ethical Considerations
Following the ethical guidelines of the base Whisper model:
- Should not be used to transcribe recordings without consent
- Not recommended for "subjective classification" tasks
- Should undergo robust evaluation before deployment in high-risk contexts
- May show performance variations across different languages and demographics
## 📜 License
This model is released under the **MIT License**, allowing for commercial and non-commercial use with proper attribution.
## 📖 Citation
If you use this model in your research or applications, please cite both our work and the original Whisper paper:
```bibtex
@misc{whisper-to-oliver,
author = {{Olib AI}},
title = {Whisper to Oliver: Fine-tuned Whisper for Real-World Conversational Audio},
year = {2024},
publisher = {Hugging Face},
howpublished = {\url{https://huggingface.co/olib-ai/whisper-to-oliver}},
}
@misc{radford2022whisper,
doi = {10.48550/ARXIV.2212.04356},
url = {https://arxiv.org/abs/2212.04356},
author = {Radford, Alec and Kim, Jong Wook and Xu, Tao and Brockman, Greg and McLeavey, Christine and Sutskever, Ilya},
title = {Robust Speech Recognition via Large-Scale Weak Supervision},
publisher = {arXiv},
year = {2022},
copyright = {arXiv.org perpetual, non-exclusive license}
}
```
## 👥 About Olib AI
[Olib AI](https://www.olib.ai) specializes in building AI services for businesses. Our team focuses on creating practical AI solutions that solve real-world problems.
**Contact Us:**
- 🌐 Website: [www.olib.ai](https://www.olib.ai)
- 📧 Akram H. Sharkar: [akram@olib.ai](mailto:akram@olib.ai)
- 📧 Maya M. Sharkar: [maya@olib.ai](mailto:maya@olib.ai)
- 💻 GitHub: [https://github.com/Olib-AI](https://github.com/Olib-AI)
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