Coursera Course: Introduction to Programming 👩‍💻 with MATLAB ~by Vanderbilt University 🎓

Matlab scripts accompanying the book "An Introduction to Audio Content Analysis" (www.AudioContentAnalysis.org)

🐳 Scripps Whale Acoustics Lab 🌎 Scripps Acoustic Ecology Lab - Triton with remoras in development

A MATLAB app for sines-transient-noise decomposition of audio signals.

This repo contains the ENF-WHU audio recording dataset collected around Wuhan University campus and the MATLAB programs for electronic network frequency (ENF) detection, enhancement, and robust estimation, in ENF-based audio forensic applications.

This project will walk you through the importance of Fast Fourier Transform (FFT) which is one of the major computation techniques in the world of Digital Signal Processing (DSP). It also explains how 'Filter Design Toolbox' can be made use of in MATLAB to design desired filters on the go.

Introductory class on Digital Signal Processing.

Objective measures of speech quality SNR

Implementation of an algorithm to detect acoustic feedback from a audio file

🔊 A Matlab application which makes the Audio Plot (Amplitude vs Time) and Fast Fourier Transform Plot (FFT) for a given sound file.

Convert a mono channel recording into binaural playback with headphones and loudspeakers

This is the accompanying repository for the article Algorithms for audio inpainting based on probabilistic nonnegative matrix factorization authored by Ondřej Mokrý, Paul Magron, Thomas Oberlin and Cédric Févotte, published in Elsevier Signal Processing.

Voice Activity Detection in audio signals using 2 wavelet-based methods (Matlab)

Matlab GUI app to resample audio files

Cochlear implant signal processing

Fast convolution and deconvolution functions using Fast Fourier Transform (FFT)

A Graphic Equaliser written in MATLAB

Build a cross-talk canceler and a speech recognizer

In this project, a simple watermarking system for spread spectrum audio is developed using the FFT transformation to insert the data into the signal host. The system is designed to resist the problem of desynchronization using synchronization codes.

Post-grad speech, audio processing & recognition coursework - source-filter LPC synthesiser and report. Achieved 95%