Ian McLoughlin

Publications

• Improving GANs for Speech Enhancement
• An End-to-End Multi-Standard OFDM Transceiver Architecture Using FPGA Partial Reconfiguration
• Spectrally efficient emission mask shaping for OFDM cognitive radios
• Effective Exploitation of Posterior Information for Attention-Based Speech Recognition
• Continuous robust sound event classification using time-frequency features and deep learning
• In-Ear Electrode EEG for Practical SSVEP BCI
• A new variance-based approach for discriminative feature extraction in machine hearing classification using spectrogram features
• Speech reconstruction using a deep partially supervised neural network.
• A New Time–Frequency Attention Tensor Network for Language Identification
• Source-Aware Context Network for Single-Channel Multi-Speaker Speech Separation
• Fisher vector based CNN architecture for image classification
• LID-Senones and Their Statistics for Language Identification
• Listening and Grouping: An Online Autoregressive Approach for Monaural Speech Separation
• Towards More Accurate Automatic Sleep Staging via Deep Transfer Learning
• End-to-end DNN-CNN classification for language identification
• Bag-of-features models based on C-DNN network for acoustic scene classification
• Beyond equal-length snippets: How long is sufficient to recognize an audio scene?
• Spatio-temporal attention pooling for audio scene classification
• What makes audio event detection harder than classification?
• Enabling Early Audio Event Detection with Neural Networks
• Deep neural network for robust speech recognition with auxiliary features from laser-Doppler vibrometer sensor
• GFM-VOC: A real-time voice quality modification system
• An attention pooling based representation learning method for speech emotion recognition
• Acoustic modeling with densely connected residual network for multichannel speech recognition
• Unifying Isolated and Overlapping Audio Event Detection with Multi-label Multi-task Convolutional Recurrent Neural Networks
• A new time-frequency attention mechanism for TDNN and CNN-LSTM-TDNN, with application to language identification
• A Region Based Attention Method for Weakly Supervised Sound Event Detection and Classification
• An improved deep embedding learning method for short duration speaker verification
• A training-based speech regeneration approach with cascading mapping models
• An effective deep embedding learning architecture for speaker verification
• Improving aggregation and loss function for better embedding learning in end-to-end speaker verification system
• A robust framework for acoustic scene classification
• Formant smoothing for quality improvement of post-laryngectomised speech reconstruction
• A Spectral Glottal Flow Model for Source-filter Separation of Speech
• A Conditional Generative Model for Speech Enhancement
• Improved Conditional Generative Adversarial Net Classification for Spoken Language Recognition
• On the mental fatigue analysis of SSVEP entrainment
• Early detection of continuous and partial audio events using CNN
• Glottal Flow Synthesis for Whisper-to-Speech Conversion
• End-to-End Language Identification Using High-Order Utterance Representation with Bilinear Pooling
• Improvements on Deep Bottleneck Network based I-Vector Representation for Spoken Language Identification
• LID-senone Extraction via Deep Neural Networks for End-to-End Language Identification
• Image classification with CNN-based Fisher vector coding
• Robust Sound Event Detection in Continuous Audio Environments
• Time–Frequency Feature Fusion for Noise Robust Audio Event Classification
• Robust acoustic scene classification using a multi-spectrogram encoder-decoder framework
• Variance Normalised Features for Language and Dialect Discrimination
• Incandescent Bulb and LED Brake Lights: Novel Analysis of Reaction Times
• CNN-MoE Based Framework for Classification of Respiratory Anomalies and Lung Disease Detection
• Multi-Granularity Sequence Alignment Mapping for Encoder-Decoder Based End-to-End ASR
• The Use of Low-Frequency Ultrasound for Voice Activity Detection
• Mouth State Detection From Low-Frequency Ultrasonic Reflection
• Super-audible Voice Activity Detection
• Cross-layer TCP/IP Segmentation, Re-routing and Adaptive Modulation Techniques to Exploit Instantaneous BER Variations on Parallel Subchannels
• Reconstruction of pitch for whisper-to-speech conversion of Chinese
• Local Coding based Matching Kernel Method for Image Classification
• Deep Bottleneck Features for Spoken Language Identification
• Tone confusion in spoken and whispered Mandarin Chinese
• Task-aware Deep Bottleneck Features for Spoken Language Identification
• Shaping Spectral Leakage for IEEE 802.11p Vehicular Communications
• A new mechanical index for gauging the human bio-effects of low frequency ultrasound
• A Unified Framework for GPS Code and Carrier-Phase Multipath Mitigation Using Support Vector Regression
• Speech Playback Geometry for Smart Homes
• Whisper-to-speech conversion using restricted Boltzmann machine arrays
• Square-Rich Fixed Point Polynomial Evaluation on FPGAs
• Robust and Efficient OFDM Synchronisation for FPGA-Based Radios
• Super-Audible Voice Activity Detection
• Local coding based matching kernel method for image classification.
• Deep bottleneck features for spoken language identification.
• Mouth State Detection From Low-Frequency Ultrasonic Reflection
• Classifying watermelon ripeness by analysing acoustic signals using mobile devices
• Low frequency ultrasonic voice activity detection using convolutional neural networks
• Deep bottleneck network based i-vector representation for language identification
• The use of low-frequency ultrasound for voice activity detection
• Task-aware deep bottleneck features for spoken language identification
• Phonated speech reconstruction using twin mapping models
• Robust sound event recognition using convolutional neural networks
• Whisper-to-speech conversion using restricted Boltzmann machine arrays
• Reconstruction of phonated speech from whispers using formant-derived plausible pitch modulation
• Improved i-Vector Representation for Speaker Diarization
• Learning compact structural representations for audio events using regressor banks
• Efficient integer frequency offset estimation architecture for enhanced OFDM synchronization
• Compact convolutional neural network transfer learning for small-scale image classification
• Performance evaluation of deep bottleneck features for spoken language identification
• A spectral based visual matching method for image classification
• Comparative whisper vowel space for Singapore English and British English accents
• Robust sound event classification using deep neural networks
• Cross-layer TCP/IP segmentation, re-routing and adaptive modulation techniques to exploit instantaneous BER variations on parallel subchannels
• Efficient Large Integer Squarers on FPGA
• Reconstruction of pitch for whisper-to-speech conversion of Chinese
• Shaping Spectral Leakage for IEEE 802.11p Vehicular Communications
• Multi-task deep neural network acoustic models with model adaptation using discriminative speaker identity for whisper recognition
• Efficient multi-standard cognitive radios on FPGAs
• Tone confusion in spoken and whispered Mandarin Chinese
• Speech and Audio Processing
• Deep Bottleneck Feature for Image Classification
• Improved language identification using deep bottleneck network
• Square-rich fixed point polynomial evaluation on FPGAs
• Robust and Efficient OFDM Synchronization for FPGA-Based Radios
• Speech playback geometry for smart homes
• Human Mouth State Detection Using Low Frequency Ultrasound
• Computer Architecture: an embedded approach
• MetroBuzz: Interactive 3D visualization of spatiotemporal data
• Performance Analysis of Adaptive Modulation and Transmit Antenna Selection with Channel Prediction Errors and Feedback Delay
• Bio-effects and safety of low-intensity, low-frequency ultrasonic exposure
• Campus Mobility for the Future: The Electric Bicycle
• Channel prediction in non-regenerative multi-antenna relay selection systems
• Multi-touch Wall Displays for Informational and Interactive Collaborative Space
• Mobile Communications Using Source-Selected Multi-Antenna AF Relays Over Dual-Hop Nakagami-m Channels
• Method and apparatus for determining mouth state using low frequency ultrasonics
• Message from WEISS-12 Workshop Chairs
• Cross-layer MIMO-Links Exploiting Packet Re-routing Mechanisms and Adaptive Modulation in Diverse Channel Condition
• TCP-based Multi Parallel Links Exploiting Packet Re-routing Mechanisms in Diverse Channel Condition
• Fourier transform-based scalable image quality measure
• Classifying watermelon ripeness by analysing acoustic signals using mobile devices
• Reconstruction of continuous voiced speech from whispers
• Efficient Squarer Design for FPGA Implementation
• A Comprehensive Vowel Space for Whispered Speech
• Exemplar Based Language Recognition Method For Short-Duration Speech Segments
• Measuring resonances of the vocal tract using frequency sweeps at the lips
• GPS multipath mitigation: a nonlinear regression approach
• Packet Switching, AM Adjustment and Retry Mechanisms for Cross-Layer MIMO Link Design
• A new mechanical index for gauging the human bioeffects of low frequency ultrasound.
• Human mouth state detection using low frequency ultrasound
• Modelling WSNS using OMNeT++
• From whispers to normal speech: Offering natural voice to laryngectomees
• Cross-layer MIMO-link exploiting packet re-routing mechanisms and adaptive modulation in diverse channel condition
• TCP-based multi parallel links exploiting packet re-routing mechanisms in diverse channel condition
• Reconstruction of continuous voiced speech from whispers
• Multi-touch wall displays for informational and interactive collaborative space
• Multi-touch wall displays for informational and interactive collaborative space
• Performance analysis of adaptive modulation and transmit antenna selection with channel prediction errors and feedback delay
• Open-Source and Consumer Electronics—The Back Door to World Domination: Why Reinvent the Wheel (Especially When Nice People are Giving Wheels Away for Free)?
• Virtualized Development and Testing for Embedded Cluster Computing
• Bio-effects and safety of low-intensity, low-frequency ultrasonic exposure
• Message from WEISS-12 workshop chairs
• Multi-touch wall displays for informational and interactive collaborative space
• Mobile communications using source-selected multi-antenna AF relays over dual-hop nakagami- m channels
• A comprehensive vowel space for whispered speech
• Measuring resonances of the vocal tract using frequency sweeps at the lips
• MetroBuzz: Interactive 3D visualization of spatiotemporal data
• Channel prediction in non-regenerative multi-antenna relay selection systems
• Fourier transform-based scalable image quality measure
• GPS multipath mitigation: A nonlinear regression approach
• Artificial phonation for patients suffering voice box lesions
• Nonintrusive quality assessment of noise suppressed speech with Mel-filtered energies and support vector regression
• TCP/IP link layer error mitigation for MIMO wireless links
• Whisper vowel diagrams for Singapore English
• Data collection, communications and processing in the Sumatran GPS array (SuGAr)
• Low-power correlation for IEEE 802.16 OFDM synchronization on FPGA
• An embedded systems graduate education for Singapore
• Method,Apparatus and Computer Readable Medium for Fast Arithmetic in Digital Logic
• Linear predictive analysis for ultrasonic speech
• Applied Speech and Audio Processing
• Autoregressive Modelling for Linear Prediction of Ultrasonic Speech
• Data collection, communications and processing in the Sumatran GPS array (SuGAr)
• Effects of Channel Prediction for Transmit Antenna Selection With Maximal-Ratio Combining in Rayleigh Fading
• Development of Nano-satellite Space segment and Ground Station
• Method and System for Reconstructing Speech from an Input Signal Comprising Whispers
• Message from the honorary chair
• Message from the APESER 2010 Program Chairs
• Analysis-by-synthesis method for whisper-speech reconstruction
• Dynamic scaling scheme for hardware-accelerated edge detection
• Fast and accurate GCPS selection scheme for SAR image registration based on an improved Trajkovic corner detector
• Joint audio video quality evaluation for distance or online education systems
• Regeneration of Speech in Voice-Loss Patients
• Applied speech and audio processing: With matlab ® examples
• Vowel Intelligibility in Chinese
• A group ring construction of the extended binary Golay code
• Speech Recognition for Smart Homes
• Fast and accurate GCPS selection scheme for SAR image registration based on an improved Trajkovic corner detector
• Performance of Dual-Hop Multi-Antenna Systems with Fixed Gain Amplify-and-Forward Relay Selection
• SLC SAR speckle filtering using homoskedastic features in logarithmic transformed domain
• Spectral enhancement of whispered speech based on probability mass function
• Effects of channel prediction for transmit antenna selection with maximal-ratio combining in Rayleigh fading
• FMLE SAR speckle filter using the distance consistency property in homoskedasticlog-transformed domain
• Virtualized development and testing of embedded computing clusters
• A perspective on the experiential learning of computer architecture
• Autoregressive modelling for linear prediction of ultrasonic speech
• Linear predictive analysis for ultrasonic speech
• Message from the APESER 2010 program chairs
• Neural network-assisted reconstruction of full polarimetric SAR information
• Point-to-point OMNeT++ based simulation of reliable transmission using realistic segmentation and reassembly with error control
• Reconstruction of normal sounding speech for laryngectomy patients through a modified CELP codec
• Reliability through redundant parallelism for micro-satellite computing
• Voiced speech from whispers for post-laryngectomised patients
• Speech rehabilitation methods for laryngectomised patients
• Toward a comprehensive vowel space for whispered speech
• Channel prediction for mitigating feedback link issues in transmit antenna selection systems
• Non-intrusive speech quality assessment with support vector regression
• Hardware-accelerated edge detection for polarimetric synthetic aperture radar data
• Joint audio video quality evaluation for distance or online education systems
• Keypress biometrics for user validation in mobile consumer devices
• Secure embedded systems: The threat of reverse engineering
• Analysis-by-synthesis method for whisper-speech reconstruction
• Predictive receive-directed antenna selection for quasistatic Rayleigh fading channel
• An embedded systems graduate education for Singapore
• Linux as a teaching aid for embedded systems
• Tone discrimination in Mandarin Chinese
• Achieving low-cost high-reliability computation through redundant parallel processing
• FPGA implementation of space-time block coding systems
• Channel estimation complexity reduction using caching
• First beowulf cluster in space
• Centralised computation service architecture for the X-Sat micro-satellite
• Design, testing and verification of a microsatellite on-board data processing unit using commercial grade processors
• Error Detection and Correction for microsatellite software running on \ COTS\ processors
• Achieving low-cost high-reliability computation through redundant parallel processing
• Extension of proposal of standards for intelligibility tests of Chinese speech: CDRT-tone
• Fault tolerance through redundant \ COTS\ components for satellite processing applications
• A methodology for improving PESQ accuracy for Chinese speech
• Method and Apparatus for Speech Enhancement in a Speech Communication System
• Method and Apparatus for Fast Arithmetic in Digital Logic
• How to track pitch pulses in LP residual? Joint time-frequency distribution approach
• How to track pitch pulses? - Joint time-frequency distribution approach
• Adaptive bit allocation for LSP parameter quantization
• Fault-tolerant Computer for Low Earth Orbit Micro-satellites
• Embedded Linux platform for a fault tolerant space based parallel computer
• Evaluation of ITU-T G. 728 as a voice over IP codec for Chinese speech
• Fault tolerant computer for low Earth orbit micro satellites
• Low-cost space-borne processing on a reconfigurable parallel architecture
• Link layer error mitigation in rural UHF-MIMO linking systems
• Low Complexity Detection Algorithms For A MIMO-OFDM System
• Intelligibility evaluation of GSM coder for Mandarin speech using CDRT
• A scalable parallel computational core for embedded processing
• A methodology for improving \ PESQ\ accuracy for \ C\ hinese speech
• How to track pitch pulse in LP residual - Joint Time-Frequency Distribution Approach
• A modular computational engine for communications processing
• Intelligibility evaluation of GSM coder for Mandarin speech using CDRT
• Joint Time- Frequency Distribution Analysis of Pitch Pulses
• A reconfigurable platform for \ MIMO\ research - realtime implementation of a 4x4 adaptive multi-variate \ DFE\
• Improvements Relating to Radio Communication Systems
• Time reversal space time block coding with channel estimation errors
• The 2k2: A modular computational toolkit for embedded signal processing
• Fault tolerance through redundant COTS components for satellite processing applications
• Channel estimation complexity reduction using caching
• Transmit antenna selection for UHF MIMO linking
• A methodology for improving PESQ accuracy for Chinese speech
• A scalable parallel computational core for embedded processing
• Perceptual audio data concealment and watermarking scheme using direct frequency domain substitution
• Link layer error mitigation in rural UHF-MIMO linking systems
• An FPGA-based MIMO and space-time processing platform
• Performance investigation and implementation of a real-time adaptive MIMO-DFE system
• The art of public speaking for engineers
• FPGA implementation of space-time block coding systems
• Fault tolerant computer for low earth orbit micro satellites
• Low-cost space-borne processing on a reconfigurable parallel architecture
• Extension of proposal of standards for intelligibility tests of Chinese speech: CDRT-tone
• Intelligibility evaluation of GSM coder for Mandarin speech using CDRT
• How to track pitch pulses in LP residual? - Joint Time-Frequency Distribution approach
• Adaptive bit allocation for LSP parameter quantization
• Speech recognition engine adaptions for smart home dialogues
• LSP parameter interpretation for speech classification
• Method and Apparatus for Speech Enhancement in a Speech Communications System
• Adaptive bit allocation for LSP parameter quantization
• Subjective Intelligibility Testing of Chinese Speech
• Adaptive bit allocation for LSP\ parameter quantization
• DSP software development
• Mandarin speech coding using a modified RPE-LTP technique
• Line spectral pairs
• MP3 player on JUMPtec DIMM-PC/486-I
• Data concealment in audio using a nonlinear frequency distribution of PRBS coded data and frequency-domain LSB insertion.
• Hardware architecture for data concealment using sub-band coding, LSB coding and pseudo-random bit stream generators
• Hardware architecture for LSB data concealment using subband filterbank coding
• Hardware architecture for data concealment using sub-band coding, LSB coding and pseudo-random bit stream generators
• Hardware architecture for data concealment using sub-band coding, LSB coding and pseudo-random bit stream generators
• Proposal of standards for intelligibility testing of Chinese Speech
• Variable Rate Coding Techniques for Mandarin Speech over Packet Networks
• Switched-basis LSP quantization
• Mandarin speech coding using a modified RPE_LTP technique
• Novel dynamic bit allocation method for LSP quantization
• Proposal of standards for intelligibility tests of Chinese speech
• Performance comparison of ICA neural networks separating audio signals
• The implementation of a high accuracy optical measurement technique of particular interest to miniature optics
• LSP-based speech modification for intelligibility enhancement
• THE IMPLEMENTATION OF A HIGH-ACCURACY OPTICAL MEASUREMENT TECHNIQUE OF PARTICULAR INTEREST TO MINIATURE OPTICS
• Hardware architecture for data concealment using sub-band coding, LSB coding and pseudo-random bit stream generators
• LSP analysis and processing for speech coders
• LSP analysis and processing for speech coders
• Data concealment in audio using a nonlinear frequency distribution of PRBS coded data and frequency-domain LSB insertion
• High accuracy displacement measurements for integrated optical components
• Analysis of adaptive modulation with antenna selection under channel prediction errors
• Predictive Transmit Antenna Selection with Maximal Ratio Combining
• Reverse engineering of embedded consumer electronic systems
• LSP-based speech modification for intelligibility enhancement
• Mandarin Speech Coding using a Modified RPE_LTP Technique
• Novel dynamic bit allocation method for LSP quantization
• Hardware architecture for data concealment using sub-band coding, LSB coding and pseudo-random bit stream generators
• The implementation of a high accuracy optical measurement technique of particular interest to miniature optics
• Modelling WSNs Using OMNeT++
• Hardware architecture for data concealment using sub-band coding, LSB coding and pseudo-random bit stream generators
• Mandarin speech coding using a modified RPE_LTP technique
• Novel dynamic bit allocation method for LSP quantization
• Data concealment in audio using a nonlinear frequency distribution of PRBS coded data and frequency-domain LSB insertion
• Proposal of standards for intelligibility tests of Chinese speech
• LSP analysis and processing for speech coders
• LSP-based speech modification for intelligibility enhancement
• Packet Switching, AM Adjustment and Retry Mechanisms for Cross-Layer MIMO Link Design
• Investigating the Cognitive Response of Brake Lights in Initiating Braking Action Using EEG
• Cross-Lingual Self-training to Learn Multilingual Representation for Low-Resource Speech Recognition
• DOMAIN ROBUST DEEP EMBEDDING LEARNING FOR SPEAKER RECOGNITION
• Wearable and implantable medical devices A fantastic voyage
• Tragus based vagus nerve stimulation for stress reduction
• JOINT GENERATIVE-CONTRASTIVE REPRESENTATION LEARNING FOR ANOMALOUS SOUND DETECTION
• SELF-SUPERVISED REPRESENTATION LEARNING FOR UNSUPERVISED ANOMALOUS SOUND DETECTION UNDER DOMAIN SHIFT
• Extremely low footprint end-to-end ASR system for smart device
• An effective speaker recognition method based on joint identification and verification supervisions
• Non-Verbal Auditory Aspects of Human-Service Robot Interaction
• FRONTEND ATTRIBUTES DISENTANGLEMENT FOR SPEECH EMOTION RECOGNITION
• Acoustic Based Footstep Detection in Pervasive Healthcare
• On the potential of transauricular electrical stimulation to reduce visually induced motion sickness
• Automated Assessment of Glottal Dysfunction Through Unified Acoustic Voice Analysis
• Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
• An Ensemble of Deep Learning Frameworks for Predicting Respiratory Anomalies
• Auditory evoked potential detection during pure-tone audiometry
• Multi-view audio and music classification
• On the potential of transcutaneous auricular vagus nerve stimulation to reduce visually induced motion sickness
• An effective deep embedding learning method based on dense-residual networks for speaker verification
• Paraformer: Fast and Accurate Parallel Transformer for Non-autoregressive End-to-End Speech Recognition
• AST-SED: AN EFFECTIVE SOUND EVENT DETECTION METHOD BASED ON AUDIO SPECTROGRAM TRANSFORMER
• Inception-Based Network and Multi-Spectrogram Ensemble Applied to Predict Respiratory Anomalies and Lung Diseases
• Vibro-motor Reprocessing Therapy towards Managing Motion Sickness Reduction: Evidence from EEG
• An effective perturbation based semi-supervised learning method for sound event detection
• Motion sickness reduction through vibro-motor reprocessing therapy: A first study
• Automatic assessment of dysarthric severity level using audio-video cross-modal approach in deep learning
• Auricular Vagus Nerve Stimulation for Stress Reduction: Evidence from Alpha Prefrontal Asymmetry
• Multi-view audio and music classification
• A weight moving average based alternate decoupled learning algorithm for long-tailed language identification
• Robust Prototype Learning for Anomalous Sound Detection
• Fine-tuning Audio Spectrogram Transformer with Task-aware Adapters for Sound Event Detection
• D-MONA: A dilated mixed-order non-local attention network for speaker and language recognition
• An effective mutual mean teaching based domain adaptation method for sound event detection
• An improved mean teacher based method for large scale weakly labeled semi-supervised sound event detection
• Class-Aware Distribution Alignment based Unsupervised Domain Adaptation for Speaker Verification
• An Ensemble of Deep Learning Frameworks Applied For Predicting Respiratory Anomalies
• Self-attention generative adversarial network for speech enhancement
• A Light-weight Deep Learning Model for Remote Sensing Image Classification
• An Online Speaker-aware Speech Separation Approach Based on Time-domain Representation
• Extremely low footprint end-to-end ASR system for smart device
• On the Use of a Spectral Glottal Model for the Source-filter Separation of Speech
• Enabling early audio event detection with neural networks
• Semi-supervised end-to-end ASR via teacher-student learning with conditional posterior distribution
• Incandescent bulb and LED brake lights: Novel analysis of reaction times
• Task-Aware Mean Teacher Method for Large Scale Weakly Labeled Semi-Supervised Sound Event Detection
• Inception-based network and multi-spectrogram ensemble applied for predicting respiratory anomalies and lung diseases
• On multitask loss function for audio event detection and localization
• Robust Deep Learning Framework for Predicting Respiratory Anomalies and Diseases
• Improving GANs for speech enhancement
• Self-attention generative adversarial network for speech enhancement
• SAN-M: Memory Equipped Self-Attention for End-to-End Speech Recognition
• Triplet-Center Loss Based Deep Embedding Learning Method for Speaker Verification
• Robust deep learning framework for predicting respiratory anomalies and diseases
• Towards more accurate automatic sleep staging via deep transfer learning
• Unifying isolated and overlapping audio event detection with multi-label multi-task convolutional recurrent neural networks
• SAN-M: Memory equipped self-attention for end-to-end speech recognition
• Deep Feature Embedding and Hierarchical Classification for Audio Scene Classification
• Deep feature embedding and hierarchical classification for audio scene classification
• CNN-MoE based framework for classification of respiratory anomalies and lung disease detection
• LSTM-TDNN WITH CONVOLUTIONAL FRONT-END FOR DIALECT IDENTIFICATION IN THE 2019 MULTI-GENRE BROADCAST CHALLENGE
• A Framework for Understanding the Weaponisation of the Internet
• Proceedings of the 9th IRC Conference on Science, Engineering, and Technology
• Proceedings of the 9th IRC Conference on Science, Engineering, and Technology
• Interactive Dance with Visual Background Recomposition using BLE Beacons
• Interactive Dance with Visual Background Recomposition using BLE Beacons