slaq-ai-engine / app.py
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"""
Enhanced Stuttering Detection API
==================================
FastAPI backend with adaptive, research-based stuttering detection.
No hardcoded thresholds - uses statistical methods (Modified Z-Score/MAD).
Improvements over previous version:
- Adaptive thresholding using Modified Z-Score (Median Absolute Deviation)
- Multi-feature acoustic analysis (RMS, Pitch, MFCCs, Spectral features)
- Speaking-rate normalization for accurate severity assessment
- Detection of 5 dysfluency types with confidence scores
- Research-backed algorithms from recent stuttering detection literature
"""
import os
import torch
import librosa
import numpy as np
from fastapi import FastAPI, File, UploadFile, Form, HTTPException
from fastapi.responses import JSONResponse
from transformers import Wav2Vec2ForCTC, AutoProcessor
from typing import Optional, Dict, List
import traceback
from scipy import signal
app = FastAPI(
title="SLAQ Enhanced AI Engine",
description="Adaptive stuttering detection with multi-feature analysis",
version="2.0.0"
)
# --- CONFIGURATION ---
ASR_MODEL_ID = "facebook/mms-1b-all"
SAMPLE_RATE = 16000
print("πŸ”„ Loading Enhanced AI Models...")
# Load ASR Model for transcription
try:
processor = AutoProcessor.from_pretrained(ASR_MODEL_ID)
model = Wav2Vec2ForCTC.from_pretrained(ASR_MODEL_ID)
print(f"βœ… ASR Model loaded: {ASR_MODEL_ID}")
except Exception as e:
print(f"❌ Failed to load ASR model: {e}")
raise e
# Language Mapping
LANG_MAP = {
'hindi': 'hin', 'tamil': 'tam', 'telugu': 'tel', 'marathi': 'mar',
'bengali': 'ben', 'gujarati': 'guj', 'kannada': 'kan', 'malayalam': 'mal',
'punjabi': 'pan', 'urdu': 'urd', 'english': 'eng', 'auto': 'auto'
}
class EnhancedStutterDetector:
"""
Enhanced stuttering detection using adaptive statistical methods.
Based on recent research (2023-2025) in dysfluency detection.
"""
def __init__(self, sample_rate: int = 16000):
self.sr = sample_rate
self.mad_threshold = 3.5 # Modified Z-Score threshold
def analyze(self, y: np.ndarray, sr: int) -> Dict:
"""Main analysis pipeline."""
duration = len(y) / sr
# Extract multi-dimensional acoustic features
features = self._extract_features(y, sr)
# Detect speaking rate
speaking_rate = self._estimate_speaking_rate(y, sr)
# Detect dysfluency events
events = []
events.extend(self._detect_blocks(y, sr, features))
events.extend(self._detect_prolongations(y, sr, features))
events.extend(self._detect_sound_repetitions(y, sr, features))
events.extend(self._detect_word_repetitions(y, sr, features))
events.extend(self._detect_interjections(y, sr, features))
# Sort by time
events.sort(key=lambda x: x['start'])
# Calculate adaptive severity
severity_score = self._calculate_severity(events, duration, speaking_rate)
return {
'events': events,
'total_events': len(events),
'severity_score': severity_score,
'severity_label': self._get_severity_label(severity_score),
'speaking_rate': speaking_rate,
'duration': duration,
'event_counts': self._count_types(events)
}
def _extract_features(self, y: np.ndarray, sr: int) -> Dict:
"""Extract acoustic features."""
frame_length = int(0.025 * sr)
hop_length = int(0.010 * sr)
features = {}
# Energy (RMS)
rms = librosa.feature.rms(y=y, frame_length=frame_length, hop_length=hop_length)[0]
features['rms'] = rms
# Pitch (F0)
f0 = librosa.yin(y, fmin=librosa.note_to_hz('C2'), fmax=librosa.note_to_hz('C7'), sr=sr)
features['f0'] = f0
# Spectral features
features['spectral_centroid'] = librosa.feature.spectral_centroid(y=y, sr=sr, hop_length=hop_length)[0]
features['spectral_rolloff'] = librosa.feature.spectral_rolloff(y=y, sr=sr, hop_length=hop_length)[0]
features['zcr'] = librosa.feature.zero_crossing_rate(y, frame_length=frame_length, hop_length=hop_length)[0]
# MFCCs
mfcc = librosa.feature.mfcc(y=y, sr=sr, n_mfcc=13, hop_length=hop_length)
features['mfcc'] = mfcc
# Time mapping
features['hop_length'] = hop_length
features['frame_times'] = librosa.frames_to_time(np.arange(len(rms)), sr=sr, hop_length=hop_length)
return features
def _estimate_speaking_rate(self, y: np.ndarray, sr: int) -> float:
"""Estimate speaking rate (syllables/sec)."""
onset_env = librosa.onset.onset_strength(y=y, sr=sr)
onsets = librosa.onset.onset_detect(onset_envelope=onset_env, sr=sr, units='time')
duration = len(y) / sr
return len(onsets) / duration if duration > 0 else 0.0
def _modified_z_score(self, data: np.ndarray) -> np.ndarray:
"""Calculate Modified Z-Score using MAD (more robust than standard Z-score)."""
median = np.median(data)
mad = np.median(np.abs(data - median))
if mad < 1e-10:
mad = np.mean(np.abs(data - median))
if mad < 1e-10:
return np.zeros_like(data)
return 0.6745 * (data - median) / mad
def _detect_blocks(self, y: np.ndarray, sr: int, features: Dict) -> List[Dict]:
"""Detect blocks (abnormal silent pauses)."""
rms = features['rms']
frame_times = features['frame_times']
# Adaptive silence threshold using Modified Z-Score
rms_z = self._modified_z_score(rms)
is_silent = rms_z < -self.mad_threshold
blocks = []
in_block = False
block_start = 0
for i, silent in enumerate(is_silent):
if silent and not in_block:
block_start = frame_times[i]
in_block = True
elif not silent and in_block:
block_end = frame_times[i]
duration = block_end - block_start
if 0.2 < duration < 2.0:
blocks.append({
'type': 'block',
'start': float(block_start),
'end': float(block_end),
'duration': float(duration),
'confidence': float(np.mean(np.abs(rms_z[max(0, i-10):i])))
})
in_block = False
return blocks
def _detect_prolongations(self, y: np.ndarray, sr: int, features: Dict) -> List[Dict]:
"""Detect prolongations (stable sound segments)."""
rms = features['rms']
f0 = features['f0']
frame_times = features['frame_times']
prolongations = []
window = 20
for i in range(window, len(rms) - window):
win_rms = rms[i-window:i+window]
win_f0 = f0[i-window:i+window]
rms_cv = np.std(win_rms) / (np.mean(win_rms) + 1e-10)
f0_cv = np.std(win_f0) / (np.mean(win_f0) + 1e-10)
if rms_cv < 0.1 and f0_cv < 0.15 and np.mean(win_rms) > np.median(rms) * 0.3:
if prolongations and frame_times[i] - prolongations[-1]['end'] < 0.1:
prolongations[-1]['end'] = float(frame_times[i])
prolongations[-1]['duration'] = prolongations[-1]['end'] - prolongations[-1]['start']
else:
start = frame_times[max(0, i-window)]
end = frame_times[min(len(frame_times)-1, i+window)]
prolongations.append({
'type': 'prolongation',
'start': float(start),
'end': float(end),
'duration': float(end - start),
'confidence': float(1.0 - (rms_cv + f0_cv) / 2)
})
return [p for p in prolongations if 0.3 < p['duration'] < 3.0]
def _detect_sound_repetitions(self, y: np.ndarray, sr: int, features: Dict) -> List[Dict]:
"""Detect sound repetitions using spectral similarity."""
mfcc = features['mfcc']
frame_times = features['frame_times']
repetitions = []
window = 15
for i in range(window, len(frame_times) - window * 2):
curr = mfcc[:, i:i+window].flatten()
next = mfcc[:, i+window:i+2*window].flatten()
if len(curr) > 0 and len(next) > 0:
similarity = np.dot(curr, next) / (np.linalg.norm(curr) * np.linalg.norm(next) + 1e-10)
if similarity > 0.85:
start = frame_times[i]
end = frame_times[min(len(frame_times)-1, i+2*window)]
repetitions.append({
'type': 'sound_repetition',
'start': float(start),
'end': float(end),
'duration': float(end - start),
'confidence': float(similarity)
})
return [r for r in repetitions if 0.1 < r['duration'] < 1.5]
def _detect_word_repetitions(self, y: np.ndarray, sr: int, features: Dict) -> List[Dict]:
"""Detect word repetitions using autocorrelation."""
rms = features['rms']
frame_times = features['frame_times']
rms_norm = (rms - np.mean(rms)) / (np.std(rms) + 1e-10)
autocorr = np.correlate(rms_norm, rms_norm, mode='full')
autocorr = autocorr[len(autocorr)//2:]
word_window = 30
peaks, _ = signal.find_peaks(
autocorr[word_window:word_window*3],
height=np.percentile(autocorr, 75),
distance=word_window//2
)
repetitions = []
for peak in peaks:
idx = peak + word_window
if idx < len(frame_times):
start = frame_times[max(0, idx-word_window)]
end = frame_times[min(len(frame_times)-1, idx+word_window)]
repetitions.append({
'type': 'word_repetition',
'start': float(start),
'end': float(end),
'duration': float(end - start),
'confidence': 0.7
})
return [r for r in repetitions if 0.3 < r['duration'] < 2.0]
def _detect_interjections(self, y: np.ndarray, sr: int, features: Dict) -> List[Dict]:
"""Detect interjections (um, uh, ah)."""
rms = features['rms']
centroid = features['spectral_centroid']
frame_times = features['frame_times']
centroid_z = self._modified_z_score(centroid)
unusual = np.abs(centroid_z) > self.mad_threshold
interjections = []
in_interj = False
start_idx = 0
for i, is_unusual in enumerate(unusual):
if is_unusual and rms[i] > np.median(rms) * 0.2:
if not in_interj:
start_idx = i
in_interj = True
elif in_interj:
duration = (i - start_idx) * features['hop_length'] / sr
if 0.1 < duration < 0.8:
interjections.append({
'type': 'interjection',
'start': float(frame_times[start_idx]),
'end': float(frame_times[i]),
'duration': float(duration),
'confidence': float(np.mean(np.abs(centroid_z[start_idx:i])))
})
in_interj = False
return interjections
def _calculate_severity(self, events: List[Dict], duration: float, rate: float) -> float:
"""Calculate adaptive severity score (0-100)."""
if duration <= 0:
return 0.0
counts = self._count_types(events)
total_time = sum(e['duration'] for e in events)
# Dysfluency percentage
dysfluency_pct = (total_time / duration) * 100
# Event frequency (per minute)
event_freq = (len(events) / duration) * 60
# Weighted count (blocks/prolongations more severe)
weights = {'block': 2.0, 'prolongation': 1.8, 'sound_repetition': 1.5,
'word_repetition': 1.3, 'interjection': 1.0}
weighted = sum(counts.get(t, 0) * w for t, w in weights.items())
# Rate normalization
rate_factor = min(rate / 4.0, 2.0) if rate > 0 else 1.0
severity = (
dysfluency_pct * 0.4 +
(event_freq / rate_factor) * 0.3 +
(weighted / rate_factor) * 0.3
)
return float(np.clip(severity, 0, 100))
def _count_types(self, events: List[Dict]) -> Dict[str, int]:
"""Count events by type."""
counts = {}
for e in events:
counts[e['type']] = counts.get(e['type'], 0) + 1
return counts
def _get_severity_label(self, score: float) -> str:
"""Convert score to label."""
if score < 10: return 'none'
elif score < 25: return 'mild'
elif score < 50: return 'moderate'
elif score < 75: return 'severe'
else: return 'very_severe'
# Initialize detector
stutter_detector = EnhancedStutterDetector(sample_rate=SAMPLE_RATE)
print("βœ… Enhanced Stutter Detector initialized")
@app.get("/")
def home():
return {
"status": "running",
"service": "SLAQ Enhanced AI Engine",
"version": "2.0.0",
"features": [
"Adaptive thresholding (Modified Z-Score/MAD)",
"Multi-feature acoustic analysis",
"Speaking-rate normalization",
"5 dysfluency types detection",
"Multilingual support (MMS-1B)"
],
"model": ASR_MODEL_ID
}
@app.get("/health")
def health():
return {"status": "healthy", "model_loaded": True}
@app.post("/analyze")
async def analyze_audio(
audio: UploadFile = File(...),
transcript: Optional[str] = Form(""),
language: Optional[str] = Form("auto")
):
"""
Analyze audio for stuttering events with adaptive detection.
Args:
audio: Audio file (WAV, MP3, etc.)
transcript: Optional reference transcript for comparison
language: Language code or 'auto' for detection
Returns:
Comprehensive stuttering analysis with adaptive thresholds
"""
temp_filename = f"temp_{audio.filename}"
try:
# Save uploaded file
with open(temp_filename, "wb") as buffer:
buffer.write(await audio.read())
# Load audio
speech, sr = librosa.load(temp_filename, sr=SAMPLE_RATE)
# --- LANGUAGE HANDLING ---
lang_code = LANG_MAP.get(str(language).lower(), 'eng')
if lang_code != 'auto':
try:
processor.tokenizer.set_target_lang(lang_code)
model.load_adapter(lang_code)
except:
print(f"⚠️ Adapter not found for {lang_code}, using eng")
lang_code = 'eng'
processor.tokenizer.set_target_lang('eng')
model.load_adapter('eng')
else:
# For auto mode, default to English
lang_code = 'eng'
processor.tokenizer.set_target_lang('eng')
model.load_adapter('eng')
# --- TRANSCRIPTION ---
inputs = processor(speech, sampling_rate=SAMPLE_RATE, return_tensors="pt")
with torch.no_grad():
outputs = model(**inputs)
logits = outputs.logits
predicted_ids = torch.argmax(logits, dim=-1)
actual_transcript = processor.batch_decode(predicted_ids)[0]
confidence = float(torch.mean(torch.nn.functional.softmax(logits, dim=-1).max(dim=-1).values))
# --- ENHANCED ACOUSTIC ANALYSIS ---
analysis = stutter_detector.analyze(speech, sr)
# --- TRANSCRIPT COMPARISON (if provided) ---
mismatch_pct = 0.0
if transcript:
import Levenshtein
dist = Levenshtein.distance(actual_transcript, transcript)
mismatch_pct = (dist / max(len(transcript), 1)) * 100
# Format timestamps
timestamps = [
{
'type': evt['type'],
'start': evt['start'],
'end': evt['end'],
'duration': evt['duration'],
'confidence': evt.get('confidence', 0.5)
}
for evt in analysis['events']
]
# Calculate total stutter duration
total_stutter_duration = sum(evt['duration'] for evt in analysis['events'])
return {
"actual_transcript": actual_transcript,
"target_transcript": transcript or "",
"mismatch_percentage": round(mismatch_pct, 2),
"stutter_timestamps": timestamps,
"total_stutter_duration": round(total_stutter_duration, 2),
"stutter_frequency": analysis['total_events'],
"severity": analysis['severity_label'],
"severity_score": round(analysis['severity_score'], 2),
"confidence_score": round(confidence, 2),
"model_version": f"enhanced-adaptive-v2 ({lang_code})",
"language_detected": lang_code,
"speaking_rate": round(analysis['speaking_rate'], 2),
"event_breakdown": analysis['event_counts'],
"dysfluency_rate": round(analysis['total_events'] / (analysis['duration'] / 60), 2) if analysis['duration'] > 0 else 0
}
except Exception as e:
traceback.print_exc()
raise HTTPException(status_code=500, detail=str(e))
finally:
if os.path.exists(temp_filename):
os.remove(temp_filename)
if __name__ == "__main__":
import uvicorn
print("\nπŸš€ Starting Enhanced SLAQ AI Engine...")
print("πŸ“Š Features: Adaptive thresholds, MAD-based detection, Multi-feature analysis")
print("🌐 Access at: http://localhost:8000")
print("πŸ“– Docs at: http://localhost:8000/docs\n")
uvicorn.run(app, host="0.0.0.0", port=8000)