Jiri Schimmel

Bio: Jiri Schimmel is an academic researcher from Brno University of Technology. The author has contributed to research in topics: Audio signal processing & Digital audio. The author has an hindex of 6, co-authored 25 publications receiving 133 citations.

Real-time guitar tube amplifier simulation using an approximation of differential equations

Abstract: Digital simulation of guitar tube amplifiers is still an opened topic. The efficient implementation of several parts of the guitar amplifier is presented in this paper. This implementation is based on the pre-computation of the solution of the nonlinear differential system and further approximation of the solution. It reduces the computational complexity while the accuracy is comparable with the numerical solution. The method is used for simulation of different parts of the guitar amplifier, namely a triode preamp stage, a phase splitter and a push-pull amplifier. Finally, the results and comparison with other methods are discussed.

Psychoacoustically Motivated Audio Declipping Based on Weighted l 1 Minimization

Abstract: A novel method for audio declipping based on sparsity is presented. The method incorporates psychoacoustic information by weighting the transform coefficients in the $\ell_{1}$ minimization. Weighting leads to an improved quality of restoration while retaining a low complexity of the algorithm. Three possible constructions of the weights are proposed, based on the absolute threshold of hearing, the global masking threshold and on a quadratic curve. Experiments compare the restoration quality according to the signal-to-distortion ratio (SDR) and PEMO-Q objective difference grade (ODG) and indicate that with correctly chosen weights, the presented method is able to compete, or even outperform, the current state of the art.

Real-Time Guitar Preamp Simulation Using Modified Blockwise Method and Approximations

Abstract: The designing of algorithms for real-time digital simulation of analog effects and amplifiers brings two contradictory requirements: accuracy versus computational efficiency. In this paper, the simulation of a typical guitar tube preamp using an approximation of the solution of differential equations is discussed with regard to accuracy and computational complexity. The solution of circuit equations is precomputed and stored in N-D tables. The stored values are approximated, and therefore different approximation techniques are investigated as well. The approximated functions are used for output signal computation and also for circuit state update. The designed algorithm is compared to the numerical solution of the given preamp and also to the real preamp.

Audible Aliasing Distortion in Digital Audio Synthesis

Abstract: This paper deals with aliasing distortion in digital audio signal synthesis of classic periodic waveforms with infinite Fourier series, for electronic musical instruments. When these waveforms are generated in the digital domain then the aliasing appears due to its unlimited bandwidth. There are several techniques for the synthesis of these signals that have been designed to avoid or reduce the aliasing distortion. However, these techniques have high computing demands. One can say that today's computers have enough computing power to use these methods. However, we have to realize that today’s computer-aided music production requires tens of multi-timbre voices generated simultaneously by software synthesizers and the most of the computing power must be reserved for harddisc recording subsystem and real-time audio processing of many audio channels with a lot of audio effects. Trivially generated classic analog synthesizer waveforms are therefore still effective for sound synthesis. We cannot avoid the aliasing distortion but spectral components produced by the aliasing can be masked with harmonic components and thus made inaudible if sufficient oversampling ratio is used. This paper deals with the assessment of audible aliasing distortion with the help of a psychoacoustic model of simultaneous masking and compares the computing demands of trivial generation using oversampling with those of other methods.

Digital speech processing, synthesis, and recognition

Abstract: Principal characteristics of speech speech production models speech analysis and analysis-synthesis systems linear predictive coding (LPC) analysis speech coding speech synthesis speech recognition future directions of speech processing. Appendices: convolution and z-transform vector quantization algorithm neural nests.

数字信号处理 = Understanding digital signal processing

Abstract: From the Publisher: This is undoubtedly the most accessible book on digital signal processing (DSP) available to the beginner. Using intuitive explanations and well-chosen examples, this book gives you the tools to develop a fundamental understanding of DSP theory. The author covers the essential mathematics by explaining the meaning and significance of the key DSP equations. Comprehensive in scope, and gentle in approach, the book will help you achieve a thorough grasp of the basics and move gradually to more sophisticated DSP concepts and applications.

A review of digital techniques for modeling vacuum-tube guitar amplifiers

Abstract: Although semiconductor technologies have displaced vacuum-tube devices in nearly all fields of electronics, vacuum tubes are still widely used in professional guitar amplifiers. A major reason for this is that electric-guitar amplifiers are typically overdriven, that is, operated in such a way that the output saturates. Vacuum tubes distort the signal in a different manner compared to solid-state electronics, and human listeners tend to prefer this. This might be because the distinctive tone of tube amplifiers was popularized in the 1950s and 1960s by early rock and roll bands, so musicians and listeners have become accustomed to tube distortion. Some studies on the perceptual aspects of vacuum-tube and solid-state distortion have been published (e.g., Hamm 1973; Bussey and Haigler 1981; Santo 1994). Despite their acclaimed tone, vacuum-tube amplifiers have certain shortcomings: large size and weight, poor durability, high power consumption, high price, and often poor availability of spare parts. Thus, it is not surprising that many attempts have been made to emulate guitar tube amplifiers using smaller and cheaper solid-state analog circuits (e.g., Todokoro 1976; Sondermeyer 1984). The next step in the evolution of tube-amplifier emulation has been to simulate the amplifiers using computers and digital signal processors (DSP). A primary advantage of digital emulation is that the same hardware can be used for modeling many different tube amplifiers and effects. When a new model is to be added, new parameter values or program code are simply uploaded to the device. Furthermore, amplifier models can be implemented

Wavelet Maxima Dispersion for Breathy to Tense Voice Discrimination

Abstract: This paper proposes a new parameter, the Maxima Dispersion Quotient (MDQ), for differentiating breathy to tense voice. Maxima derived following wavelet decomposition are often used for detecting edges in image processing, where locations of these maxima organize in the vicinity of the edge location. Similarly for tense voice, which typically displays sharp glottal closing characteristics, maxima following wavelet analysis are organized in the vicinity of the glottal closure instant (GCI). Contrastingly, as the phonation type tends away from tense voice towards a breathier phonation it is observed that the maxima become increasingly dispersed. The MDQ parameter is designed to quantify the extent of this dispersion and is shown to compare favorably to existing voice quality parameters, particularly for the analysis of continuous speech. Also, classification experiments reveal a significant improvement in the detection of the voice qualities when MDQ is included as an input to the classifier. Finally, MDQ is shown to be robust to additive noise down to a Signal-to-Noise Ratio of 10 dB.