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Update app.py
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app.py
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import gradio as gr
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import warnings
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import torch
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from transformers import WhisperTokenizer, WhisperForConditionalGeneration, WhisperProcessor
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import soundfile as sf
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import ffmpeg
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import os
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from
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from huggingface_hub import InferenceClient
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from gradio_client import Client, file
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import spaces
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import time
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warnings.filterwarnings("ignore")
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# Set up device
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device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
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torch_dtype = torch.float32
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# Move model to device
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model.to(device)
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def convert_audio_format(audio_path):
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ffmpeg.input(audio_path).output(output_path, format='wav', ar='16000').run(overwrite_output=True)
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return output_path
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# @spaces.GPU(duration=120, queue=False)
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def transcribe_audio(audio_file, batch_size=4):
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start_time = time.time()
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audio_path = convert_audio_format(audio_file)
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audio_input, sample_rate = sf.read(audio_path)
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chunk_size = 16000 *
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chunks = [audio_input[i:i + chunk_size] for i in range(0, len(audio_input), chunk_size)]
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transcription = ""
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inputs.input_features,
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max_length=2048,
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num_beams=7,
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attention_mask=attention_mask
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)
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transcription += " ".join(processor.batch_decode(output, skip_special_tokens=True)) + " "
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result = f"Transcription: {transcription.strip()}\n\nTime taken: {transcription_time:.2f} seconds\nNumber of words: {word_count}"
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return result
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# HTML
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banner_html = """
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"""
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image_html = """
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<div style="text-align: center; margin-top: 20px;">
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<img src="https://huggingface.co/spaces/camparchimedes/ola_s-audioshop/
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</div>
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"""
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gr.Markdown("# ππ―π’ππ’π ππππ ππΌπΎπ¦Ύβ‘ @{NbAiLab/whisper-norwegian-medium}\nUpload audio file: β")
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audio_input = gr.Audio(type="filepath")
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batch_size_input = gr.Slider(minimum=1, maximum=16, step=1, label="Batch Size")
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transcription_output = gr.Textbox()
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transcribe_button = gr.Button("Transcribe")
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# Launch interface
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iface.launch(share=True, debug=True)
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import gradio as gr
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import warnings
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import torch
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#from transformers import WhisperTokenizer, WhisperForConditionalGeneration, WhisperProcessor
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from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq
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import soundfile as sf
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import ffmpeg
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import os
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from fpdf import FPDF
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import time
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from sklearn.feature_extraction.text import TfidfVectorizer
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from sklearn.cluster import KMeans
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import re
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import nltk
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from nltk.tokenize import word_tokenize
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from nltk.corpus import stopwords
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import pandas as pd
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warnings.filterwarnings("ignore")
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nltk.download('punkt')
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nltk.download('stopwords')
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#tokenizer = WhisperTokenizer.from_pretrained("NbAiLabBeta/nb-whisper-large")
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#model = WhisperForConditionalGeneration.from_pretrained("NbAiLabBeta/nb-whisper-large")
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#processor = WhisperProcessor.from_pretrained("NbAiLabBeta/nb-whisper-large")
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generation_config = {
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"temperature": 0.8,
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"top_p": 0.9,
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"top_k": 0.5,
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"max_output_tokens": 2048
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}
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processor = AutoProcessor.from_pretrained("NbAiLabBeta/nb-whisper-large-semantic")
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model = AutoModelForSpeechSeq2Seq.from_pretrained("NbAiLabBeta/nb-whisper-large-semantic")
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device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
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torch_dtype = torch.float32
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model.to(device)
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def convert_audio_format(audio_path):
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ffmpeg.input(audio_path).output(output_path, format='wav', ar='16000').run(overwrite_output=True)
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return output_path
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def transcribe_audio(audio_file, batch_size=4):
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start_time = time.time()
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audio_path = convert_audio_format(audio_file)
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audio_input, sample_rate = sf.read(audio_path)
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chunk_size = 16000 * 30
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chunks = [audio_input[i:i + chunk_size] for i in range(0, len(audio_input), chunk_size)]
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transcription = ""
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inputs.input_features,
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max_length=2048,
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num_beams=7,
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task="transcribe",
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attention_mask=attention_mask
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transcription += " ".join(processor.batch_decode(output, skip_special_tokens=True)) + " "
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result = f"Transcription: {transcription.strip()}\n\nTime taken: {transcription_time:.2f} seconds\nNumber of words: {word_count}"
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return transcription.strip(), result
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def save_to_pdf(transcription):
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pdf = FPDF()
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pdf.add_page()
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pdf.set_font("Arial", size=12)
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pdf.multi_cell(0, 10, transcription)
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pdf_output_path = "transcription.pdf"
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pdf.output(pdf_output_path)
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return pdf_output_path
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def summarize_text(transcription):
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sentences = transcription.split(". ")
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vectorizer = TfidfVectorizer(stop_words='norwegian')
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X = vectorizer.fit_transform(sentences)
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kmeans = KMeans(n_clusters=1)
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kmeans.fit(X)
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avg = X.mean(axis=0)
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summary = [sentences[i] for i in kmeans.predict(avg)]
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return ". ".join(summary) + "."
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# HTML
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banner_html = """
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"""
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image_html = """
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<div style="text-align: center; margin-top: 20px;">
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<img src="https://huggingface.co/spaces/camparchimedes/ola_s-audioshop/raw/main/500x_picture.png" alt="picture" width="50%" height="auto">
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</div>
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"""
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gr.Markdown("# ππ―π’ππ’π ππππ ππΌπΎπ¦Ύβ‘ @{NbAiLab/whisper-norwegian-medium}\nUpload audio file: β")
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audio_input = gr.Audio(type="filepath")
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batch_size_input = gr.Slider(minimum=1, maximum=16, step=1, label="Batch Size")
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transcription_output = gr.Textbox(label="Transcription")
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pdf_output = gr.File(label="Download Transcription as PDF")
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summary_output = gr.Textbox(label="Summary")
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transcribe_button = gr.Button("Transcribe")
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def process_audio(audio_file, batch_size):
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transcription, result = transcribe_audio(audio_file, batch_size)
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pdf_path = save_to_pdf(transcription)
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summary = summarize_text(transcription)
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return result, pdf_path, summary
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transcribe_button.click(fn=process_audio, inputs=[audio_input, batch_size_input], outputs=[transcription_output, pdf_output, summary_output])
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# Launch interface
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iface.launch(share=True, debug=True)
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