Research
This is a compilation of a few papers I've published. A lot of this work was done in collaboration with great people! Special thanks to them
The current project describes the design of a tuning or alignment processor for a system based on multiple satellite speakers and a single subwoofer. It explains the methodology used to solve this problem and the procedure to arrive to a viable solution. On one side, the procedure was based on filter design techniques to optimize phase relationships; however, these phase relations are often unknown due to the possibility of changing the relative positions of the speakers. Accordingly, a statistical analysis was used to determine the most stable set of parameters across different speaker locations and acoustical environments, implying that the same alignment parameters can be implemented in multiple circumstances without significant performance degradation.
SpeakerDrum is an instrument composed of multiple Dual Voice Coil speakers (DVC) where two coils are used to drive the same membrane. However, in this case, one of them is used as a microphone which is then used by the performer as an input interface of percussive gestures. Of course, this leads to poten- tial feedback, but with enough control, a compelling exploration of resonance haptic feedback and sound embodiment is possible.
This project intends to do an analysis of different spatialization algorithms in order to develop multiple alternative methods to simulate apparent positions of sound phenomena. Nevertheless, beyond the research development, this project represents an effort to clearly merge different types of knowledge such as mathematics, physics, signal and system analysis, programming languages, etc... pointing towards an artistic objective. At last the reader will encounter a detailed description behind the development of a suite of algorithms presented as plugins for a single purpose: to pursue a burst of possi- bilities in mixing engineering regarding the manipulation of stereo imaging.
Filter Digitization through the Bilinear Transformation is often considered a very good all-around method to produce equalizer sections. The method is well behaved in terms of stability and ease of implementation; however, the frequency warping produced by the transformation leads to abnormalities near the Nyquist frequency. Moreover, it is impossible to design parametric sections whose analog center frequencies are defined above the Nyquist frequency. These filters, even with center frequencies outside of the hearing range, have effects that extend into the hearing bandwidth with desirable characteristics during mixing and mastering.
Surpassing these limitations, while controlling the abnormalities of the warping produced by the Bilinear Transform through an alternative definition of the Bilinear constant, is the purpose of this paper. In the process, also a correction factor is discussed for the bandwidth of the parametric section to correct abnormalities affecting the digitization of this parameter.
This paper was an exploration of different possibilities of making a laptop a physical instrument. It was developed in the framework of SLOrK (Stanford Laptop Orchestra) and it came to produce two great performances at CCRMA and at Bing Concert Hall at Stanford
This paper describes a method to create all pass filters with controllable shapes comparable to parametric or shelving filters but for the phase response. The method produces FIR filters but these can be heavily optimized depending on how the Hilbert transform is implemented.
This paper describes how to create crossovers with flat magnitude with both chebyshev and elliptic filters
Fourier Paradoxes
This is not at all a common paper, it is more a reflection of how Fourier theory can lead to unintuitive conclusions. Because of that it is not intended to present new information since most of these things have been said by others before, but to condense in a simple explanation all of these ideas
This paper is about measuring aliasing byproduct in compression algorithms and how we can tweak Fourier theory to understand compression as a convolution of spectra. Therefore, aliasing can be computed simply by how much of the spectrum extends beyond Nyquist. Also algorithms are shared on how to measure aliasing in compressors with some routing tricks
This paper is about techniques, considerations and workflows in music field recordings