created app.py
Browse filescreated

app.py
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import streamlit as st
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from transformers import pipeline
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import torchaudio
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import tempfile
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import os
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import torch
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# Create a Streamlit app title
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st.title("ASR with Hugging Face Whisper")
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# Load the ASR model
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asr = pipeline(task = "automatic-speech-recognition", model="openai/whisper-large-v2",
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device=0 if torch.cuda.is_available() else "cpu")
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# Create a file uploader widget
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uploaded_audio = st.file_uploader("Upload an audio file (wav/mp3)")
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# Check if an audio file is uploaded
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if uploaded_audio:
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# Read the uploaded audio file
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audio_data, sample_rate = torchaudio.load(uploaded_audio)
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# Perform ASR on the uploaded audio
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with st.spinner("Performing ASR..."):
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transcriptions = asr(audio_data.numpy(), sample_rate=sample_rate)
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# Display the ASR result
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st.subheader("Transcription:")
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for idx, transcription in enumerate(transcriptions):
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st.write(f"Segment {idx + 1}: {transcription['text']}")
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# Provide instructions
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st.write("Instructions:")
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st.write("1. Upload an audio file in WAV or MP3 format.")
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st.write("2. Click the 'Perform ASR' button to transcribe the audio.")
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# Add a sample audio file for testing (optional)
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st.write("Sample Audio for Testing:")
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sample_audio = "Wave_files_demos_Welcome.wav"
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st.audio(sample_audio, format="audio/wav")
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# Define the path to the sample audio file
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sample_audio_path = os.path.join(os.getcwd(), sample_audio)
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# Add a button to transcribe the sample audio (optional)
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if st.button("Transcribe Sample Audio"):
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# Read the sample audio file
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sample_audio_data, sample_audio_rate = torchaudio.load(sample_audio_path)
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# Perform ASR on the sample audio
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with st.spinner("Performing ASR..."):
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sample_transcriptions = asr(sample_audio_data.numpy(), sample_rate=sample_audio_rate)
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# Display the ASR result for the sample audio
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st.subheader("Transcription (Sample Audio):")
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for idx, transcription in enumerate(sample_transcriptions):
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st.write(f"Segment {idx + 1}: {transcription['text']}")
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