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Update app.py
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app.py
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@@ -1,16 +1,19 @@
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import streamlit as st
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import tempfile
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import os
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# Load the
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@st.cache_resource
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def load_model():
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model = load_model()
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# Streamlit UI
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st.title("Hebrew Speech-to-Text Transcription")
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# Upload audio file
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@@ -22,14 +25,24 @@ if uploaded_file is not None:
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temp_audio.write(uploaded_file.read())
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temp_audio_path = temp_audio.name
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# Transcribe the audio
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st.write("Transcribing...")
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try:
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st.subheader("Transcription:")
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st.write(
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except Exception as e:
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st.error(f"Error: {str(e)}")
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# Clean up the temporary file
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os.remove(temp_audio_path)
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import streamlit as st
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import tempfile
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import os
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import librosa # For audio resampling
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import torch
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from transformers import WhisperProcessor, WhisperForConditionalGeneration
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# Load the model and processor
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@st.cache_resource
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def load_model():
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processor = WhisperProcessor.from_pretrained("ivrit-ai/whisper-large-v3-turbo")
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model = WhisperForConditionalGeneration.from_pretrained("ivrit-ai/whisper-large-v3-turbo")
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return processor, model
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processor, model = load_model()
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st.title("Hebrew Speech-to-Text Transcription")
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# Upload audio file
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temp_audio.write(uploaded_file.read())
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temp_audio_path = temp_audio.name
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try:
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# Load and resample audio to 16kHz (required by Whisper)
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speech_array, sampling_rate = librosa.load(temp_audio_path, sr=16000)
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# Preprocess audio
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inputs = processor(speech_array, sampling_rate=16000, return_tensors="pt")
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# Generate transcription
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with torch.no_grad():
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predicted_ids = model.generate(inputs.input_features)
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transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0]
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st.subheader("Transcription:")
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st.write(transcription)
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except Exception as e:
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st.error(f"Error: {str(e)}")
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# Clean up the temporary file
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os.remove(temp_audio_path)
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