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Update README.md

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@@ -15,44 +15,4 @@ this model trained on validation segment of data for one epoch with 0.05 loss an
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  how to use the model in colab:
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- #start
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- pip install torch torchaudio transformers librosa gradio
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- from transformers import WhisperProcessor, WhisperForConditionalGeneration
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- import torch
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-
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- #Load your fine-tuned Whisper model and processor
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- model_name = "hackergeek98/tinyyyy_whisper"
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- processor = WhisperProcessor.from_pretrained(model_name)
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- model = WhisperForConditionalGeneration.from_pretrained(model_name)
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-
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- #Force the model to transcribe in Persian
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- model.config.forced_decoder_ids = processor.get_decoder_prompt_ids(language="fa", task="transcribe")
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-
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- #Move model to GPU if available
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- device = "cuda" if torch.cuda.is_available() else "cpu"
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- model.to(device)
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- import librosa
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-
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- def transcribe_audio(audio_file):
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- # Load audio file using librosa (supports multiple formats)
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- audio_data, sampling_rate = librosa.load(audio_file, sr=16000) # Resample to 16kHz
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-
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- # Preprocess the audio
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- inputs = processor(audio_data, sampling_rate=sampling_rate, return_tensors="pt").input_features.to(device)
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-
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- # Generate transcription
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- with torch.no_grad():
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- predicted_ids = model.generate(inputs)
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-
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- # Decode the transcription
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- transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
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- return transcription
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- from google.colab import files
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-
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- #Upload an audio file
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- uploaded = files.upload()
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- audio_file = list(uploaded.keys())[0]
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-
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- #Transcribe the audio
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- transcription = transcribe_audio(audio_file)
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- print("Transcription:", transcription)
 
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  how to use the model in colab:
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