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VoxShield — Team B

Voice Scam Detection & Speaker Analysis

A multitask model on a fine-tuned WavLM backbone produces both judgments — human vs. AI-generated, and accent profile — from a single audio clip.

🔗 Try it live | 🎬 Demo video


Executive Summary

VoxShield developed an AI service platform that detects, in real time, whether a voice is generated by AI or spoken by a real person. As voice phishing and AI-generated voice scams become increasingly common, the platform provides an effective way to identify potentially fraudulent audio. In addition, when the speaker is a real person, the system can classify the speaker's accent or country of origin. This technology can help protect the general public from voice-based fraud while also providing valuable support to law enforcement and investigative agencies. Overall, VoxShield offers a practical AI solution for enhancing voice authentication and public safety.


Table of Contents


Demo

Human voice AI-generated voice
Human audio input Synthetic audio input
REAL verdict with accent result SYNTHETIC verdict
Verified as human → language & accent stages run Flagged as synthetic → accent stages skipped

Team

image

image image profile
Jinwoong Lee Hyunwoo Jung Hyejin Park Seongho Choi Geonah Lee
Data Acquisition Lead Data Systems Presentation & Design Engineering Engineering
GitHub GitHub GitHub GitHub GitHub

Role Breakdown

Name Major Role
Jinwoong Lee Robotics Engineering Data Acquisition Lead — sourced, curated, and validated the training corpora underpinning both the detection and accent models
Hyunwoo Jung Electronic Engineering Data Systems — structured and maintained dataset documentation and technical records across the pipeline
Hyejin Park Climate Environmental Engineering Presentation & Design — crafted the visual narrative and deck that brings the system's results to life
Seongho Choi Police Administration & Computer Science (Double Major) Engineering — bridges investigative domain expertise with hands-on model and system development
Geonah Lee Computer Science Engineering — architected and built the core detection and classification pipeline

Service Overview

What it does

Given a single audio clip, the service estimates two things at once:

  1. Accent (nationality) estimation — how close the speaker's English sounds to each of several accent regions, expressed as per-class probabilities. It currently covers 6 classes — US / UK / CA / AU / IN / CN — and uses the raw softmax probabilities directly as "accent closeness" percentages (e.g., US 30% · UK 45% · IN 10% · ...).
  2. AI synthetic speech (deepfake) detection — for the same input, whether the clip is a real human recording (REAL) or generated by TTS/voice cloning (FAKE).

Both tasks run through a single multitask model: a shared WavLM backbone with a country head and a real/fake head on top, so accent estimation and authenticity detection don't require two separate models.

Why we built it

  • Estimating a speaker's accent region from audio alone is a recurring need in language learning feedback, call-center quality monitoring, and speaker profiling.
  • At the same time, as AI-generated synthetic speech spreads, more use cases need to verify whether audio is actually human before trusting it. Solving both problems in one pipeline is why we chose a multitask design.

How it works

Audio input
   │
   ▼
Pretrained audio encoder (WavLM, fine-tuned)
   │
   ├──▶ Country (accent) classification head ──▶ US/UK/CA/AU/IN/CN probabilities
   │
   └──▶ Real/fake classification head        ──▶ REAL/FAKE probabilities
  • Training data lives in a GCS bucket (gcloud-first); training runs on Vertex AI Custom Jobs (T4/A100). Locally we only keep source code, environment variables, and docs.
  • Results can be checked directly through a Cloud Run demo (model tester) — upload a clip and see accent percentages plus the REAL/FAKE verdict.

Key features

  • One backbone, two heads — accent classification and synthetic-speech detection are trained jointly on a shared encoder, keeping inference cost low.
  • Frame-level evidence — frame-level logits are preserved so a time-axis accent heatmap can be rendered (infer.py --plot).
  • Domain randomization augmentation — speed, band-limiting, reverb, and colored noise augmentations simulate channel/recording differences, improving generalization to unseen corpora.
  • Reproducible training pipeline — curation → split → train → evaluate → deploy is fully scripted, with a report written for every experiment (classifier/reports/).

Tech Stack

Area Technology
Model PyTorch, HuggingFace Transformers (WavLM backbone, fine-tuned)
Training infra Google Cloud Vertex AI Custom Job (T4 / A100), Cloud Build, Artifact Registry
Data Google Cloud Storage (GCS), speaker-disjoint train/val/test splits
Serving / demo Google Cloud Run
Language Python

Docs

  • PRD.md — Product Requirements Document

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