A new method for determining optimum dimension ratios for small rectangular rooms has been presented. In a theoretical model, an exact description of the room impulse response was used. Based on the impulse response, a frequency response of a room was calculated to find changes in the sound pressure level over the frequency range 20–200 Hz. These changes depend on the source and receiver positions, thus, a new metric equivalent to an average frequency response was introduced to quantify the overall sound pressure variation within the room for a selected source position. A numerical procedure was employed to seek a minimum value of the deviation of the sound pressure level response from a smooth fitted response determined by the quadratic polynomial regression. The most smooth frequency responses were obtained when the source was located at one of the eight corners of a room. Thus, to find the best possible dimension ratios, in the numerical procedure the optimal source position was assumed. Calculation results have shown that optimum dimension ratios depend on the room volume and the sound damping inside a room, and for small and medium volumes these ratios are roughly 1 : 1.48 : 2.12, 1 : 1.4 : 1.89 and 1 : 1.2 : 1.45. When the room volume was suitably large, the ratio 1 : 1.2 : 1.44 was found to be the best one.
In this paper a concept of finite impulse response (FIR) narrow band-stop (notch) filter with non-zero initial conditions, based on infinite impulse response (IIR) prototype filter, is proposed. The filter described in this paper is used to suppress power line noise from ECG signals. In order to reduce the transient response of the proposed FIR notch filter, optimal initial conditions for the filter have been determined. The algorithm for finding the length of the initial conditions vector is presented. The proposed values of the length of initial conditions vector, for several ECG signals and interfering frequencies, are calculated. The proposed filters are tested using various ECG signals. Computer simulations demonstrate that the proposed FIR filters outperform traditional FIR filters with initial conditions set to zero.
In this work, an approach to the design of broadband thickness-mode piezoelectric transducer is pre- sented. In this approach, simulation of discrete time model of the impulse response of matched and backed piezoelectric transducer is used to design high sensitivity, broad bandwidth, and short-duration impulse response transducers. The effect of matching the performance of transmitting and receiving air backed PZT-5A transducer working into water load is studied. The optimum acoustical characteristics of the quarter wavelength matching layers are determined by a compromise between sensitivity and pulse duration. The thickness of bonding layers is smaller than that of the quarter wavelength matching layers so that they do not change the resonance peak significantly. Our calculations show that the −3 dB air backed transducer bandwidth can be improved considerably by using quarter wavelength matching layers. The computer model developed in this work to predict the behavior of multilayer structures driven by a transient waveform agrees well with measured results. Furthermore, the advantage of this this model over other approaches is that the time signal for optimum set of matching layers can be predicted rapidly
Surface Acoustic Wave (SAW) devices like delay lines, filters, resonators etc., are nowadays extensively used as principal solid state components in many electronic applications and chemical vapour sensors. To bring out the best from these SAW devices, computational design and modelling are resorted too. The present paper proposes the modelling of 400 MHz ST-X Quartz based SAW delay line, by three models namely, Impulse Response Model (IRM), Crossed-field Equivalent Circuit Model (ECM) and Couplingof- Modes (COM) model. MATLABr is employed as a computational tool to model the experimental output of the SAW device. A comparative discussion of the modelled device results is also provided.
A method for evaluating the dynamic characteristics of force transducers against small and short-duration impact forces is developed. In this method, a small mass collides with a force transducer and the impact force is measured with high accuracy as the inertial force of the mass. A pneumatic linear bearing is used to achieve linear motion with sufficiently small friction acting on the mass, which is the moving part of the bearing. Small and short-duration impact forces with a maximum impact force of approximately 5 N and minimum half-value width of approximately 1 ms are applied to a force transducer and the impulse responses are evaluated.
A theoretical method has been presented to describe sound decay in building enclosures and to simulate the room impulse response (RIR) employed for prediction of the indoor reverberation characteristics. The method was based on a solution of wave equation having the form of a series whose time-decaying components represent responses of acoustic modes to an impulse sound source. For small sound absorption on room walls this solution was found by means of the method of variation of parameters. A decay function was computed via the time-reverse integration of the squared RIR. Computer simulations carried out for a rectangular enclosure have proved that the RIR function reproduces the structure of a sound field in the initial stage of sound decay suffciently well. They have also shown that band-limitedness of the RIR has evident influence on the shape of the decay function and predicted decay times.
The model considered in the paper is defined as VAR with the prior distribution for parameters generated by the dynamic stochastic general equilibrium (DSGE) model. The degree of economic restrictions in the DSGE-VAR model is controlled by the weighting parameter. In the paper there is investigated the impact of the weighting parameter prior specifications for the posterior shape of impulse response functions (IRFs). In case of conditional models the paths of IRFs highly depend on the value of the weighting parameter that is set arbitrary. When considering full estimation with different prior types, means and gradual change in the dispersion the posterior time paths of IRFs are similar in models with high values of the marginal data density.
In virtual acoustics or artificial reverberation, impulse responses can be split so that direct and reflected components of the sound field are reproduced via separate loudspeakers. The authors had investigated the perceptual effect of angular separation of those components in commonly used 5.0 and 7.0 multichannel systems, with one and three sound sources respectively (Kleczkowski et al., 2015, J. Audio Eng. Soc. 63, 428-443). In that work, each of the front channels of the 7.0 system was fed with only one sound source. In this work a similar experiment is reported, but with phantom sound sources between the front loud- speakers. The perceptual advantage of separation was found to be more consistent than in the condition of discrete sound sources. The results were analysed both for pooled listeners and in three groups, according to experience. The advantage of separation was the highest in the group of experienced listeners.
Estimating the fundamental frequency and harmonic parameters is basic for signal modelling in a power supply system. Differing from the existing parameter estimation algorithms either in power quality monitoring or in harmonic compensation, the proposed algorithm enables a simultaneous estimation of the fundamental frequency, the amplitudes and phases of harmonic waves. A pure sinusoid is obtained from an input multiharmonic input signal by finite-impulse-response (FIR) comb filters. Proposed algorithm is based on the use of partial derivatives of the processed signal and the weighted estimation procedure to estimate the fundamental frequency, the amplitude and the phase of a multi-sinusoidal signal. The proposed algorithm can be applied in signal reconstruction, spectral estimation, system identification, as well as in other important signal processing problems. The simulation results verify the effectiveness of the proposed algorithm.
Virtual or active acoustics refers to the generation of a simulated room response by means of electroacoustics and digital signal processing. An artificial room response may include sound reflections and reverberation as well as other acoustic features mimicking the actual room. They will cause the listener to have an impression of being immersed in virtual acoustics of another simulated room that coexists with the actual physical room. Using low-latency broadband multi-channel convolution and carefully measured room data, optimized transducers for rendering of sound fields, and an intuitive touch control user interface, it is possible to achieve a very high perceived quality of active acoustics, with a straightforward adjustability. The electroacoustically coupled room resulting from such optimization does not merely produce an equivalent of a back-door reverberation chamber, but rather a fully functional complete room superimposed on the physical room, yet with highly selectable and adjustable acoustic response. The utility of such active system for music recording and performance is discussed and supported with examples.