Norbert Toma

Comparison of different experimental methods for the assessment of the room’s acoustics

The paper presents the acoustics analysis of three different enclosed spaces. These spaces (rooms) have different geometrical shapes and sizes and serve for different purposes. The early decay time, reverberation time, clarity and center time are evaluated with Dirac, WinMLS, Aurora and Caracad software using simple, low-cost equipment. The listed acoustic parameters were determined using linear sine sweep and impulsive sources. Comparisons between experimental measurements, simulations and analytic results were done. The room impulse response measurement proved to be the most reliable method to evaluate the properties of different rooms even if the measurements are perturbed by non-idealities of the low-cost equipment.

Comparative Performance Analysis of Artificial Reverberation Algorithms

The paper considers two efficient improved artificial reverberation algorithms built with an early Schroeders and a late reverberator: modified Schroeders or an absorbent all-pass. Both of them simulate the behavior of a real concert hall by taking into account the variation with frequency of the reverberation time. Several reverberation algorithms can be found in the literature, but very few design and performance analysis hints are given. Our paper includes a detailed design of the reverberators, simulations are run for each algorithm, acoustic parameters are computed and an objective performance analysis is carried out

Aspects of reverberation algorithms

Artificial reverberation is used in different domains from cinematography to psychology for enhancement of rooms acoustical properties. Reverberation is a phenomenon with benefit effect in music. The paper starts with the presentation of the reverberation phenomenon followed by a description of some early and late reverberation algorithms. Then we show the structure of two complete reverberators: classical and improved ones. The advantages of the latter with respect to the impulse response and natural sounding are pointed out.

Evaluation of All-Pass Reverberators

The paper presents the structure and the performance of signal processing algorithms that simulate natural room reverberation. It deals with three digital complete artificial reverberators, built up of Schroeders early reverberator and a late reverberator built with all-pass filters: Schroeders, Gardners or absorbent all-pass reverberator. For each of the presented reverberation algorithms acoustic parameters are computed and compared with those of a concert hall.

On improved reverberation algorithms

Artlfcial reverberation is wed in different domains fiom CinematoRaphy to psychology. Important applications are production of special efsects, enhancement of room S momtical properties or improvement of hearing problems. l3e paper starts with the presentation of some clarsical reverberation algorithms, where neither frequency dependence for the reverberation time nor a precise rules for design can be set iip. The traditional reverberators were improved by intrdirction of absorptive j?liers. The paper provides their detailed and simulations to prove their good performance.

Experimental Acoustic Evaluation of an Auditorium

The paper presents a case history: the acoustical analysis of a rectangular auditorium. The following acoustical parameters were evaluated: early decay time, reverberation time, clarity, definition, and center time. The excitation signal was linear sweep sine and additional analysis was carried out: peak-to-noise ratio, reverberation time for empty and occupied room, standard deviation of acoustical parameters, diffusion, and just noticeable differences analysis. Conclusions about room’s destination and modeling were drawn in the end.

Novel FIR implementation for acoustic signal processing

This paper proposes a novel FIR filter hardware implementation. In applications where acoustic signals are processed (acoustic echo cancellation, reverberation, digital audio effects) the required orders of the filters might be too high to fit into hardware platforms for real-time processing The number of delay elements required by traditional FIR implementations equals to the filter order. Our proposal explores the capability to process acoustic signals at much higher frequencies than the sampling rate. The resulting hardware requires two memories, one used to store the signal samples and the other one for the filter coefficients. The hardware also requires one multiplier and an accumulator. The feasibility of the proposed system is proven by the implementation of a digital reverberator on an FPGA platform.

System identification with adaptive algorithms

The paper presents the adaptive LMS algorithm used for system identification. The unknown system is modeled by a FIR filter. After the presentation of the theoretical approach, simulations in Labview and Matlab are provided showing the performance of the method.