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---
library_name: transformers
tags: [speech-to-text, transcription, Gujarati, whisper, fine-tuned]
---
# Whisper Small - Fine-tuned for Gujarati Speech-to-Text
This model is a fine-tuned version of [openai/whisper-small](https://huggingface.co/openai/whisper-small) for Gujarati transcription and translation tasks. It is capable of converting Gujarati speech into text, and since it is based on Whisper, it supports multilingual audio inputs. This fine-tuned model was specifically trained for improving performance on Gujarati speech data.
## Model Details
### Model Description
This model was fine-tuned on Gujarati speech data to improve transcription accuracy for audio recorded in Gujarati. It has been trained to handle diverse speech inputs, including variations in accents, backgrounds, and speech styles.
- **Developed by:** [BLACK]
- **Shared by:** [None]
- **Model type:** Speech-to-Text (Fine-tuned Whisper Model)
- **Language(s):** Gujarati
- **License:** Apache-2.0
- **Finetuned from model:** [openai/whisper-small](https://huggingface.co/openai/whisper-small)
## Uses
<!-- Address questions around how the model is intended to be used, including the foreseeable users of the model and those affected by the model. -->
### Direct Use
```python
import torch
import librosa
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
processor = AutoProcessor.from_pretrained("iiBLACKii/Gujarati_VDB_Fine_Tune")
model = AutoModelForSpeechSeq2Seq.from_pretrained("iiBLACKii/Gujarati_VDB_Fine_Tune")
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
model.to(device)
def preprocess_audio(file_path, sampling_rate=16000):
audio_array, sr = librosa.load(file_path, sr=None)
if sr != sampling_rate:
audio_array = librosa.resample(audio_array, orig_sr=sr, target_sr=sampling_rate)
return audio_array
def transcribe_and_translate_audio(audio_path):
audio_array = preprocess_audio(audio_path)
input_features = processor(audio_array, return_tensors="pt", sampling_rate=16000).input_features
input_features = input_features.to(device)
with torch.no_grad():
predicted_ids = model.generate(input_features, max_length=400, num_beams=5)
transcription_or_translation = processor.batch_decode(predicted_ids, skip_special_tokens=True)
return transcription_or_translation[0]
if __name__ == "__main__":
audio_file_path = "" # .wav file path
print("Transcribing and Translating audio...")
result = transcribe_and_translate_audio(audio_file_path)
print(f"Result: {result}")
```
### Using Base Model (OpenAI)
```python
import torch
import librosa
from transformers import WhisperProcessor, WhisperForConditionalGeneration, AutoConfig
repo_name = "iiBLACKii/Gujarati_VDB_Fine_Tune"
processor = WhisperProcessor.from_pretrained(repo_name)
config = AutoConfig.from_pretrained(repo_name)
model = WhisperForConditionalGeneration.from_pretrained(repo_name, config=config)
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
model.to(device)
def preprocess_audio(file_path, sampling_rate=16000):
audio_array, sr = librosa.load(file_path, sr=None)
if sr != sampling_rate:
audio_array = librosa.resample(audio_array, orig_sr=sr, target_sr=sampling_rate)
return audio_array
def transcribe_audio(audio_path):
audio_array = preprocess_audio(audio_path)
input_features = processor.feature_extractor(
audio_array, sampling_rate=16000, return_tensors="pt"
).input_features
input_features = input_features.to(device)
with torch.no_grad():
predicted_ids = model.generate(
input_features,
max_new_tokens=400,
num_beams=5,
)
transcription = processor.tokenizer.batch_decode(predicted_ids, skip_special_tokens=True)
return transcription[0]
if __name__ == "__main__":
audio_file_path = "" #.wav file path
print("Transcribing audio...")
transcription = transcribe_audio(audio_file_path)
print(f"Transcription: {transcription}")
```
### Downstream Use [optional]
<!-- This section is for the model use when fine-tuned for a task, or when plugged into a larger ecosystem/app -->
[More Information Needed]
### Out-of-Scope Use
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[More Information Needed]
## Bias, Risks, and Limitations
<!-- This section is meant to convey both technical and sociotechnical limitations. -->
[More Information Needed]
### Recommendations
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Users (both direct and downstream) should be made aware of the risks, biases and limitations of the model. More information needed for further recommendations.
## How to Get Started with the Model
Use the code below to get started with the model.
[More Information Needed]
## Training Details
### Training Data
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### Training Procedure
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#### Preprocessing [optional]
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#### Training Hyperparameters
- **Training regime:** [More Information Needed] <!--fp32, fp16 mixed precision, bf16 mixed precision, bf16 non-mixed precision, fp16 non-mixed precision, fp8 mixed precision -->
#### Speeds, Sizes, Times [optional]
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## Evaluation
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### Testing Data, Factors & Metrics
#### Testing Data
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#### Factors
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#### Metrics
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### Results
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#### Summary
## Model Examination [optional]
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## Environmental Impact
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Carbon emissions can be estimated using the [Machine Learning Impact calculator](https://mlco2.github.io/impact#compute) presented in [Lacoste et al. (2019)](https://arxiv.org/abs/1910.09700).
- **Hardware Type:** [More Information Needed]
- **Hours used:** [More Information Needed]
- **Cloud Provider:** [More Information Needed]
- **Compute Region:** [More Information Needed]
- **Carbon Emitted:** [More Information Needed]
## Technical Specifications [optional]
### Model Architecture and Objective
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### Compute Infrastructure
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#### Hardware
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#### Software
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## Citation [optional]
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**BibTeX:**
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**APA:**
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## Glossary [optional]
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## More Information [optional]
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## Model Card Authors [optional]
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## Model Card Contact
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