taham655 commited on
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ae30e6a
1 Parent(s): 97005f9

Update app.py

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  1. app.py +68 -0
app.py CHANGED
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+ import streamlit as st
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+ import os
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+ import soundfile as sf
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+ import torch
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+ from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline
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+ # Assuming you have your .env file configured with necessary API keys or configurations
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+ # load_dotenv()
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+
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+ # Initialize the model outside the main app function to load it only once
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+
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+
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+ device = "cuda:0" if torch.cuda.is_available() else "cpu"
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+ torch_dtype = torch.float16 if torch.cuda.is_available() else torch.float32
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+
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+ model_id = "distil-whisper/distil-large-v2"
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+
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+ model = AutoModelForSpeechSeq2Seq.from_pretrained(
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+ model_id, torch_dtype=torch_dtype, low_cpu_mem_usage=True, use_safetensors=True
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+ )
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+ model.to(device)
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+
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+ processor = AutoProcessor.from_pretrained(model_id)
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+
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+ pipe = pipeline(
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+ "automatic-speech-recognition",
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+ model=model,
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+ tokenizer=processor.tokenizer,
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+ feature_extractor=processor.feature_extractor,
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+ max_new_tokens=128,
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+ chunk_length_s=15,
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+ batch_size=16,
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+ torch_dtype=torch_dtype,
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+ device=device,
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+ )
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+
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+
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+
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+ def transcribe_audio(audio_file):
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+ # Save the audio file to a temporary file
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+ with open("temp_audio_file", "wb") as f:
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+ f.write(audio_file.getbuffer())
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+
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+ # Transcribe the audio file using the Whisper model
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+ result = pipe("temp_audio_file")
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+ return result["text"]
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+
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+ # Streamlit app
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+ def main():
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+ st.title('BETTER TRANSCRIBER')
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+
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+ # Audio file uploader
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+ uploaded_file = st.file_uploader("Upload an audio file", type=["wav", "mp3", "m4a", "ogg", "flac"])
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+
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+ if uploaded_file is not None:
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+ # Show a button to start the transcription process
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+ if st.button('Transcribe'):
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+ # Show a message while transcribing
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+ with st.spinner('Transcribing...'):
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+ text = transcribe_audio(uploaded_file)
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+
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+ # Show the transcription
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+ st.subheader('Transcription:')
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+ st.write(text)
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+ else:
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+ st.write('Upload an audio file to get started.')
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+
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+ if __name__ == "__main__":
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+ main()