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# app.py
import gradio as gr
import warnings
import torch
from transformers import pipeline, WhisperTokenizer, WhisperForConditionalGeneration, WhisperProcessor
warnings.filterwarnings("ignore")
# Load tokenizer and model
tokenizer = WhisperTokenizer.from_pretrained("NbAiLabBeta/nb-whisper-medium")
model = WhisperForConditionalGeneration.from_pretrained("NbAiLabBeta/nb-whisper-medium")
processor = WhisperProcessor.from_pretrained("NbAiLabBeta/nb-whisper-medium")
# Set up the device
device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
torch_dtype = torch.float32
# Initialize pipeline
asr = pipeline("automatic-speech-recognition", model=model, tokenizer=processor.tokenizer, feature_extractor=processor.feature_extractor, device=device, torch_dtype=torch_dtype)
def transcribe_audio(audio_file):
# Perform transcription
with torch.no_grad():
output = asr(audio_file, chunk_length_s=28, generate_kwargs={"num_beams": 5, "task": "transcribe", "language": "no"})
return output["text"]
# Create Gradio interface
iface = gr.Interface(
fn=transcribe_audio,
inputs=gr.Audio(type="filepath"),
outputs="text",
title="Audio Transcription App",
description="Upload an audio file to get the transcription",
theme="default",
layout="vertical",
live=False
)
from transformers import AutoProcessor, AutoModelForSpeechSeq2Seq, AutoTokenizer, AutoModelForSeq2SeqLM
from pydub import AudioSegment
import soundfile as sf
import numpy as np
import os
import nltk
from fpdf import FPDF
import time
nltk.download('punkt')
# transcription
processor = AutoProcessor.from_pretrained("NbAiLabBeta/nb-whisper-large-semantic")
transcription_model = AutoModelForSpeechSeq2Seq.from_pretrained("NbAiLabBeta/nb-whisper-large-semantic")
# summarization
summarization_tokenizer = AutoTokenizer.from_pretrained("NbAiLab/norbert-summarization")
summarization_model = AutoModelForSeq2SeqLM.from_pretrained("NbAiLab/norbert-summarization")
# setup
device = torch.device('cuda' if torch.cuda.is_available() else 'cpu')
torch_dtype = torch.float32
# move 'em
transcription_model.to(device)
summarization_model.to(device) # PS. model needs to be told to use graph-based summary method (Lexname?)
def convert_to_wav(audio_file):
audio = AudioSegment.from_file(audio_file, format="m4a")
wav_file = "temp.wav"
audio.export(wav_file, format="wav")
return wav_file
def transcribe_audio(audio_file, batch_size=4):
start_time = time.time()
# Convert .m4a to .wav
if audio_file.endswith(".m4a"):
audio_file = convert_to_wav(audio_file)
audio_input, sample_rate = sf.read(audio_file)
chunk_size = 16000 * 30
chunks = [audio_input[i:i + chunk_size] for i in range(0, len(audio_input), chunk_size)]
transcription = ""
for i in range(0, len(chunks), batch_size):
batch_chunks = chunks[i:i + batch_size]
inputs = processor(batch_chunks, sampling_rate=16000, return_tensors="pt", padding=True)
inputs = inputs.to(device)
attention_mask = inputs.attention_mask.to(device) if 'attention_mask' in inputs else None
with torch.no_grad():
output = transcription_model.generate(
inputs.input_features,
max_length=2048, # Increase max_length for longer outputs
num_beams=7,
task="transcribe",
attention_mask=attention_mask,
# forced_decoder_ids=None, # OBS! forced_decoder_ids must not be set. Just marked it out for, just in case..
language="no"
)
transcription += " ".join(processor.batch_decode(output, skip_special_tokens=True)) + " "
end_time = time.time()
transcription_time = end_time - start_time
word_count = len(transcription.split())
result = f"Transcription: {transcription.strip()}\n\nTime taken: {transcription_time:.2f} seconds\nNumber of words: {word_count}"
return transcription.strip(), result
def summarize_text(text):
inputs = summarization_tokenizer([text], max_length=1024, return_tensors="pt", truncation=True)
inputs = inputs.to(device)
summary_ids = summarization_model.generate(inputs.input_ids, num_beams=4, max_length=150, early_stopping=True)
summary = summarization_tokenizer.decode(summary_ids[0], skip_special_tokens=True)
return summary
# HTML syntax for imagery
image_html = """
<div style="text-align: center;">
<img src="https://huggingface.co/spaces/camparchimedes/ola_s-audioshop/raw/main/Olas%20AudioSwitch%20Shop.png" alt="Banner" width="87%" height="auto">
</div>
<div style="text-align: center; margin-top: 20px;">
<img src="https://huggingface.co/spaces/camparchimedes/ola_s-audioshop/raw/main/picture.jpg" alt="Additional Image" width="50%" height="auto">
</div>
"""
# Gradio UI
iface = gr.Blocks()
with iface:
gr.HTML(image_html)
gr.Markdown("# Switch Work Audio Transcription App\nUpload an audio file to get the transcription")
audio_input = gr.Audio(type="filepath")
batch_size_input = gr.Slider(minimum=1, maximum=16, step=1, default=4, label="Batch Size")
transcription_output = gr.Textbox()
summary_output = gr.Textbox()
transcribe_button = gr.Button("Transcribe and Summarize")
def transcribe_and_summarize(audio_file, batch_size):
transcription, result = transcribe_audio(audio_file, batch_size)
summary = summarize_text(transcription)
return result, summary
transcribe_button.click(fn=transcribe_and_summarize, inputs=[audio_input, batch_size_input], outputs=[transcription_output, summary_output])
def save_to_pdf(transcription, summary):
pdf = FPDF()
pdf.add_page()
pdf.set_font("Arial", size=12)
# include transcription
pdf.multi_cell(0, 10, "Transcription:\n" + transcription)
# paragraph space
pdf.ln(10)
# include summary
pdf.multi_cell(0, 10, "Summary:\n" + summary)
pdf_output_path = "transcription_summary.pdf"
pdf.output(pdf_output_path)
return pdf_output_path
# run
iface.launch(share=True, debug=True)