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Parent(s):
90bcc62
test
Browse files- app.py +48 -58
- requirements.txt +3 -3
app.py
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import
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import tempfile
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import json
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import librosa
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import numpy as np
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import soundfile as sf
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import torch
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import gradio as gr
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from transformers import pipeline
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# Load models
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'summarizer': pipeline("summarization", model="sshleifer/distilbart-cnn-12-6", device=device)
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}
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audio_data, sample_rate = librosa.load(audio_path, sr=16000) # Whisper expects 16kHz
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audio_data = audio_data.astype(np.float32)
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with tempfile.NamedTemporaryFile(delete=False, suffix='.wav') as temp_wav:
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sf.write(temp_wav.name, audio_data, sample_rate, format='WAV')
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return temp_wav.name
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temp_wav_path = load_and_convert_audio(audio_file.name)
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# Transcription
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transcription = models['transcriber'](temp_wav_path, return_timestamps=True)
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results['transcription'] = transcription['text'] if isinstance(transcription, dict) else ' '.join([chunk['text'] for chunk in transcription])
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# Summarization
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text = results['transcription']
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words = text.split()
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chunk_size = 1000
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chunks = [' '.join(words[i:i + chunk_size]) for i in range(0, len(words), chunk_size)]
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summaries = [models['summarizer'](chunk, max_length=200, min_length=50, truncation=True)[0]['summary_text'] for chunk in chunks]
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results['summary'] = ' '.join(summaries)
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os.unlink(temp_wav_path)
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return process_audio(audio)
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#
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import streamlit as st
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import tempfile
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import soundfile as sf
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from transformers import pipeline
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# Load models
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transcriber = pipeline("automatic-speech-recognition", model="openai/whisper-tiny.en", device=-1)
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summarizer = pipeline("summarization", model="sshleifer/distilbart-cnn-12-6", device=-1)
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question_generator = pipeline("text2text-generation", model="google/t5-efficient-tiny", device=-1)
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# Upload audio file
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uploaded_file = st.file_uploader("Upload Audio", type=["wav", "mp3"])
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if uploaded_file is not None:
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# Save the uploaded file to a temporary file
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with tempfile.NamedTemporaryFile(delete=False) as temp_audio_file:
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temp_audio_file.write(uploaded_file.getbuffer())
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temp_audio_path = temp_audio_file.name
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# Read the audio file using SoundFile
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try:
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# Load audio data
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audio_data, sample_rate = sf.read(temp_audio_path)
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# Transcribing audio
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lecture_text = transcriber(temp_audio_path)["text"]
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# Preprocessing data
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num_words = len(lecture_text.split())
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max_length = min(num_words, 1024) # BART model max input length is 1024 tokens
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max_length = int(max_length * 0.75) # Convert max words to approx tokens
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if max_length > 1024:
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lecture_text = lecture_text[:int(1024 / 0.75)] # Truncate to fit the model's token limit
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# Summarization
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summary = summarizer(
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lecture_text,
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max_length=1024, # DistilBART max input length is 1024 tokens
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min_length=int(max_length * 0.1),
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truncation=True
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)
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# Clean up the summary text
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if not summary[0]["summary_text"].endswith((".", "!", "?")):
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last_period_index = summary[0]["summary_text"].rfind(".")
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if last_period_index != -1:
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summary[0]["summary_text"] = summary[0]["summary_text"][:last_period_index + 1]
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# Questions Generation
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context = f"Based on the following lecture summary: {summary[0]['summary_text']}, generate some relevant practice questions."
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questions = question_generator(context, max_new_tokens=50)
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# Output
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st.write("\nSummary:\n", summary[0]["summary_text"])
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for question in questions:
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st.write(question["generated_text"]) # Output the generated questions
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except Exception as e:
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st.error(f"Error during processing: {str(e)}")
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requirements.txt
CHANGED
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@@ -1,6 +1,6 @@
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torch
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soundfile
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-
transformers
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numpy
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streamlit
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transformers
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torch
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soundfile
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numpy
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librosa
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