In digital audio processing, one of the most common tasks is reducing background noise while keeping the original signal as clear as possible. This balance is typically expressed as maximizing the Signal-to-Noise Ratio (SNR) — […]
Key Features: Done with JUCE framework by using filter processor duplicator (IIR filter, juce::dsp::ProcessorDuplicator) Response curve (logarithmic), spectrum, audio waveform, peak and RMS level meters Volume and low-high-band pass (with 0.707 Q value) and two […]
Key Features: Done with FMOD API, diagram shows class dependency and interactive class state structure State containers and segments can be set by order, loop, and randomness Interactive music class uses an open nested loop […]
Key Features: Done with JUCE framework by using filter processor duplicator (IIR filter, juce::dsp::ProcessorDuplicator) Balanced log-linear spectrum, audio waveform, peak and RMS level meters Volume and low-high-band pass filters (with 0.707 Q value) for simple […]
When mixing in stereo, one of the most overlooked but critical checks is stereo correlation — how “in phase” your left and right channels are. This single number tells you whether your mix will hold […]
The Laplace transform and z-transform are mathematical ways of breaking an impulse response into sinusoids and decaying exponentials. This is done by expressing the system’s characteristics as one complex polynomial divided by another complex polynomial. […]
Key Features: Done with JUCE framework by using filter processor duplicator (IIR filter, juce::dsp::ProcessorDuplicator) Audio thumbnail, linear spectrum and stereo correlation windows, peak and RMS level meters Volume, low pass filter, and stereo width sliders […]
This is an experimental Schroeder reverberator implementation. I applied three combo filters and then two all pass filters without any buffer. Reverb tail was added to the end of input signal in a simple way, […]
I implemented a recursive moving average filter to reduce background noise in some test audio files. It works in the time domain, which makes it faster than FFT-based algorithms when the number of points is […]
I used the spectrogram to detect attack times in a single line melody by shaping signal’s power graph. It works better with only slow passages in which the notes are separable from each other. I […]
I applied interpolation technique to the spectral peak based on fundamental frequency after FFT was taken. It is basically to find a better estimation of the peak by using the max spectral bin (of fundamental), […]
This is my implementation for some of windows used in audio applications. I chose Hamming, Blackman, and Blackman-Harris window functions. Instead of using FFT pack, I directly applied DFT since the DFT size is not […]
This is my DFT implementation to test a wav file. It only accepts mono wav files with 44100 Hz sample rate. I used 32 floating point (with max 23 mantissa) in the range -1 and […]