app.py

import os
import tarfile
import torch
import torchaudio
import numpy as np
import streamlit as st
import matplotlib.pyplot as plt
from huggingface_hub import login
from transformers import (
    AutoProcessor,
    AutoModelForSpeechSeq2Seq,
    TrainingArguments,
    Trainer,
    DataCollatorForSeq2Seq,
)
from cryptography.fernet import Fernet

# ================================
# 1️⃣ Authenticate with Hugging Face Hub (Securely)
# ================================
HF_TOKEN = os.getenv("hf_token")  

if HF_TOKEN is None:
    raise ValueError("❌ Hugging Face API token not found. Please set it in Secrets.")

login(token=HF_TOKEN)

# ================================
# 2️⃣ Load Model & Processor
# ================================
MODEL_NAME = "AqeelShafy7/AudioSangraha-Audio_to_Text"
processor = AutoProcessor.from_pretrained(MODEL_NAME)
model = AutoModelForSpeechSeq2Seq.from_pretrained(MODEL_NAME)

device = "cuda" if torch.cuda.is_available() else "cpu"
model.to(device)
print(f"✅ Model loaded on {device}")

# ================================
# 3️⃣ Load Dataset (From Extracted Folder)
# ================================
DATASET_TAR_PATH = "dev-clean.tar.gz"
EXTRACT_PATH = "./librispeech_dev_clean"

if not os.path.exists(EXTRACT_PATH):
    print("🔄 Extracting dataset...")
    with tarfile.open(DATASET_TAR_PATH, "r:gz") as tar:
        tar.extractall(EXTRACT_PATH)
    print("✅ Extraction complete.")
else:
    print("✅ Dataset already extracted.")

AUDIO_FOLDER = os.path.join(EXTRACT_PATH, "LibriSpeech", "dev-clean")

def find_audio_files(base_folder):
    return [os.path.join(root, file)
            for root, _, files in os.walk(base_folder)
            for file in files if file.endswith(".flac")]

audio_files = find_audio_files(AUDIO_FOLDER)

if not audio_files:
    raise FileNotFoundError(f"❌ No .flac files found in {AUDIO_FOLDER}. Check dataset structure!")

print(f"✅ Found {len(audio_files)} audio files in dataset!")

# ================================
# 4️⃣ Load Transcripts
# ================================
def load_transcripts():
    transcript_dict = {}
    for root, _, files in os.walk(AUDIO_FOLDER):
        for file in files:
            if file.endswith(".txt"):
                with open(os.path.join(root, file), "r", encoding="utf-8") as f:
                    for line in f:
                        parts = line.strip().split(" ", 1)
                        if len(parts) == 2:
                            file_id, text = parts
                            transcript_dict[file_id] = text
    return transcript_dict

transcripts = load_transcripts()
if not transcripts:
    raise FileNotFoundError("❌ No transcripts found! Check dataset structure.")

print(f"✅ Loaded {len(transcripts)} transcripts.")

# ================================
# 5️⃣ Preprocess Dataset (Fixing `input_ids` issue)
# ================================
def load_and_process_audio(audio_path):
    waveform, sample_rate = torchaudio.load(audio_path)
    waveform = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(waveform)
    waveform = waveform.to(dtype=torch.float32)
    input_features = processor(waveform.squeeze().numpy(), sampling_rate=16000, return_tensors="pt").input_features[0]
    return input_features

dataset = [
    {
        "input_features": load_and_process_audio(audio_file),
        "labels": processor.tokenizer(transcripts[os.path.basename(audio_file).replace(".flac", "")],
                                      padding="max_length", truncation=True, return_tensors="pt").input_ids[0]
    }
    for audio_file in audio_files[:100] if os.path.basename(audio_file).replace(".flac", "") in transcripts
]

train_size = int(0.8 * len(dataset))
train_dataset, eval_dataset = dataset[:train_size], dataset[train_size:]

print(f"✅ Dataset Prepared! Training: {len(train_dataset)}, Evaluation: {len(eval_dataset)}")

# ================================
# 6️⃣ Streamlit UI: Fine-Tuning Hyperparameter Selection
# ================================
st.sidebar.title("🔧 Fine-Tuning Hyperparameters")
num_epochs = st.sidebar.slider("Epochs", min_value=1, max_value=10, value=3)
learning_rate = st.sidebar.select_slider("Learning Rate", options=[5e-4, 1e-4, 5e-5, 1e-5], value=5e-5)
batch_size = st.sidebar.select_slider("Batch Size", options=[2, 4, 8, 16], value=8)
attack_strength = st.sidebar.slider("Attack Strength", 0.0, 0.9, 0.1)

# ================================
# 7️⃣ Streamlit ASR Web App (Fast Decoding & Security Features)
# ================================
st.title("🎙️ Speech-to-Text ASR Model with Security Features 🎶")

audio_file = st.file_uploader("Upload an audio file", type=["wav", "mp3", "flac"])

if audio_file:
    audio_path = "temp_audio.wav"
    with open(audio_path, "wb") as f:
        f.write(audio_file.read())

    waveform, sample_rate = torchaudio.load(audio_path)
    waveform = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(waveform)
    waveform = waveform.to(dtype=torch.float32)
    
    # Simulate an adversarial attack by injecting random noise
    adversarial_waveform = waveform + (attack_strength * torch.randn_like(waveform))
    adversarial_waveform = torch.clamp(adversarial_waveform, -1.0, 1.0)
    
    input_features = processor(adversarial_waveform.squeeze().numpy(), sampling_rate=16000, return_tensors="pt").input_features.to(device)
    
    with torch.inference_mode():
        generated_ids = model.generate(input_features, max_length=200, num_beams=2, do_sample=False, use_cache=True,
                                       attention_mask=torch.ones(input_features.shape, dtype=torch.long).to(device))
        transcription = processor.batch_decode(generated_ids, skip_special_tokens=True)[0]
    
    if attack_strength > 0.1:
        st.warning("⚠️ Adversarial attack detected! Transcription may be affected.")
    
    st.success("📄 Secure Transcription:")
    st.write(transcription)