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9cb7f63 | 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 | import os
import contextlib
import wave
import librosa
import numpy as np
import pandas as pd
import parselmouth
import soundfile as sf
import webrtcvad
from tensorflow.keras.models import load_model
import joblib
import warnings
import tempfile
import json
# --- Streamlit Imports ---
import streamlit as st
from sklearn.preprocessing import StandardScaler
# --- Configuration ---
TARGET_SR = 16000
MODEL_PATH = "vocal_model.h5"
# We now use the JSON file for the scaler
SCALER_PATH_JSON = "vocal_scaler.json"
FEATURES_PATH = "feature_names.joblib"
# --- Suppress Warnings ---
warnings.filterwarnings('ignore')
os.environ['TF_CPP_MIN_LOG_LEVEL'] = '3'
# --- Caching the Models and Scaler ---
# This is a key Streamlit feature. It loads the models only once, making the app fast.
@st.cache_resource
def load_models_and_scaler():
"""Load and cache the model, scaler, and feature names."""
model = load_model(MODEL_PATH)
# Load scaler from JSON
with open(SCALER_PATH_JSON, 'r') as f:
scaler_data = json.load(f)
scaler = StandardScaler()
scaler.mean_ = np.array(scaler_data['mean_'])
scaler.scale_ = np.array(scaler_data['scale_'])
feature_names = joblib.load(FEATURES_PATH)
return model, scaler, feature_names
# --- Feature Extraction Functions (Your original functions) ---
# ... (Copy ALL your feature extraction functions here, exactly as they were) ...
def preprocess_audio(input_path, target_sr=TARGET_SR):
try:
data, sr = librosa.load(input_path, sr=None, mono=False)
if data.ndim > 1: data = data.mean(axis=0)
if sr != target_sr: data = librosa.resample(data, orig_sr=sr, target_sr=target_sr)
base, ext = os.path.splitext(input_path)
output_path = f"{base}_processed_for_prediction.wav"
sf.write(output_path, data, target_sr, subtype='PCM_16')
return output_path
except Exception as e:
st.error(f"Error preprocessing audio: {e}")
return None
def extract_features(file_path):
try:
y, sr = librosa.load(file_path, sr=None)
duration = librosa.get_duration(y=y, sr=sr)
mfccs = librosa.feature.mfcc(y=y, sr=sr, n_mfcc=13)
mfcc_means = np.mean(mfccs, axis=1)
snd = parselmouth.Sound(file_path)
pitch = snd.to_pitch()
pitch_values = pitch.selected_array['frequency']
pitch_values = pitch_values[pitch_values != 0]
pitch_mean = np.mean(pitch_values) if len(pitch_values) > 0 else 0
pitch_std = np.std(pitch_values) if len(pitch_values) > 0 else 0
point_process = parselmouth.praat.call(snd, "To PointProcess (periodic, cc)", 75, 500)
jitter_local = parselmouth.praat.call(point_process, "Get jitter (local)", 0, 0, 0.0001, 0.02, 1.3)
shimmer_local = parselmouth.praat.call([snd, point_process], "Get shimmer (local)", 0, 0, 0.0001, 0.02, 1.3, 1.6)
def read_wave(path):
with contextlib.closing(wave.open(path, 'rb')) as wf:
pcm_data, sample_rate = wf.readframes(wf.getnframes()), wf.getframerate()
return pcm_data, sample_rate
def frame_generator(frame_duration_ms, audio, sample_rate):
n = int(sample_rate * (frame_duration_ms / 1000.0) * 2)
offset = 0
while offset + n < len(audio):
yield audio[offset:offset + n]
offset += n
vad = webrtcvad.Vad(1)
audio, sample_rate = read_wave(file_path)
frames = list(frame_generator(30, audio, sample_rate))
voiced_seconds = 0
num_segments = 0
if frames:
for frame in frames:
if vad.is_speech(frame, sample_rate):
voiced_seconds += 0.03 # 30ms frame
num_segments +=1
silence_ratio = max(0, (duration - voiced_seconds) / duration) if duration > 0 else 0
speaking_rate = num_segments / duration if duration > 0 else 0
features = {
'Duration': duration,
'Pitch_Mean': pitch_mean,
'Pitch_Std': pitch_std,
'Jitter': jitter_local,
'Shimmer': shimmer_local,
'Speaking_Rate': speaking_rate,
'Silence_Ratio': silence_ratio,
}
for idx, val in enumerate(mfcc_means):
features[f'MFCC_{idx+1}'] = val
return features
except Exception as e:
st.error(f"Error extracting features: {e}")
return None
# --- Main Prediction Logic (Refactored for Streamlit) ---
def predict(audio_file_path, model, scaler, feature_names):
"""Takes an audio file path and returns the prediction results."""
processed_path = preprocess_audio(audio_file_path)
if not processed_path:
return None, None
features_dict = extract_features(processed_path)
os.remove(processed_path) # Clean up the processed file
if not features_dict:
return None, None
# Convert to DataFrame and ensure correct column order
feature_df = pd.DataFrame([features_dict])
feature_df = feature_df[feature_names]
# Scale the features
scaled_features = scaler.transform(feature_df)
# Make prediction
prediction_prob = model.predict(scaled_features, verbose=0)[0][0]
return prediction_prob, features_dict
# --- Streamlit App Interface ---
st.set_page_config(page_title="Parkinson's Voice Detector", page_icon="🩺", layout="centered")
st.title("🩺 Parkinson's Disease Detection from Voice")
st.markdown("""
This app uses a machine learning model to predict the likelihood of Parkinson's disease based on vocal features.
Upload a short voice recording (e.g., of someone saying "ahhh" for a few seconds) to get a prediction.
**Disclaimer:** This is a demonstration tool and not a substitute for professional medical advice.
""")
# Load models
try:
model, scaler, feature_names = load_models_and_scaler()
st.sidebar.success("Model and components loaded successfully!")
except Exception as e:
st.error(f"Error loading model components: {e}")
st.stop() # Stop the app if models can't be loaded
# File Uploader
uploaded_file = st.file_uploader(
"Choose a voice recording...",
type=["wav", "mp3", "ogg", "flac"]
)
if uploaded_file is not None:
st.audio(uploaded_file, format='audio/wav')
# When the user clicks the button, start prediction
if st.button("Analyze Audio", type="primary"):
# Save the uploaded file to a temporary location
with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as tmp_file:
tmp_file.write(uploaded_file.getvalue())
tmp_file_path = tmp_file.name
with st.spinner('Analyzing audio... This may take a moment.'):
try:
prediction_prob, features = predict(tmp_file_path, model, scaler, feature_names)
if prediction_prob is not None:
# Display results
st.subheader("Analysis Result")
is_parkinsons = prediction_prob > 0.5
if is_parkinsons:
st.warning(f"**Parkinson's Detected** (Confidence: {prediction_prob:.2%})")
else:
st.success(f"**Healthy** (Confidence of being healthy: {(1-prediction_prob):.2%})")
# Display confidence as a progress bar
st.progress(prediction_prob)
st.markdown(f"The model's confidence score for the presence of Parkinson's is **{prediction_prob:.2%}**.")
# Show extracted features in an expander
with st.expander("View Extracted Vocal Features"):
st.json(features)
else:
st.error("Could not process the audio file. Please try a different file.")
except Exception as e:
st.error(f"An unexpected error occurred during analysis: {e}")
finally:
# Clean up the temporary file
os.remove(tmp_file_path) |