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README.md
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license: apache-2.0
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---
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license: apache-2.0
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language:
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- uz
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- en
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- ru
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metrics:
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- wer
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base_model:
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- openai/whisper-small
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pipeline_tag: automatic-speech-recognition
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tags:
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- speech-recognition
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- whisper
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- multilingual
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- uzbek
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- russian
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- english
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---
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# Multilingual Whisper (Uz/En/Ru) — Fine-tuned Speech-to-Text Model
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A fine-tuned **Whisper Small** model optimized to transcribe **Uzbek, English, and Russian equally well**.
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This model is intended for real-world speech transcription with a balanced multilingual dataset and performs competitively against strong open-source and commercial STT solutions.
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---
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## Model Details
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### Model Description
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This model extends **OpenAI Whisper Small** by fine-tuning it on a multilingual speech mixture, aimed to deliver robust ASR performance for **Uzbek**, **English**, and **Russian** speakers.
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The goal was to reduce the performance gap between languages, especially improving **Uzbek** speech recognition, where public ASR resources are scarce.
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- **Model type:** Automatic Speech Recognition (ASR)
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- **Language(s):** Uzbek 🇺🇿, English 🇬🇧, Russian 🇷🇺
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- **License:** Apache-2.0
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- **Finetuned from:** openai/whisper-small
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- **Intended usage:** Real-time & offline speech-to-text
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---
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## Trained datasets:
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- DavronSherbaev/uzbekvoice-filtered
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- telegram-voice-messages (private collection)
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- navaistt-open-datasets
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- sovaai/russian-audiobooks
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- librispeech
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## Evaluation
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### Word Error Rate (WER) Comparison
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| Model | WER ↓ |
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|--------------------------------|----------|
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| Whisper-small-uz-v1 | **34.00%** |
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| Gemini (Commercial) | 36.21% |
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| NavaiSTT v2 (Open-Source) | 35.14% |
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| Aisha STT (Commercial) | 41.71% |
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The model **outperforms both commercial and open-source Uzbek STT models**, showing strong generalization for informal real-world speech.
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---
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## Usage Example
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```python
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from transformers import WhisperProcessor, WhisperForConditionalGeneration
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import torch
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import torchaudio
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model_id = "Firdavs222/whisper-small-uz-v1" # replace with real model repo
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processor = WhisperProcessor.from_pretrained(model_id)
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model = WhisperForConditionalGeneration.from_pretrained(model_id)
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audio, sr = torchaudio.load("audio.wav")
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inputs = processor(audio, sampling_rate=sr, return_tensors="pt")
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with torch.no_grad():
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predicted_ids = model.generate(inputs.input_features)
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text = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
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print(text) # → transcribed text here
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