Download Discrete-time Models for Non-linear Audio Systems A variety of computational models have been proposed for digital simulation of nonlinear systems with memory [1, 2, 3, 4]. They are dealing with different aspects of the problem, like methods for identification, avoiding aliasing and fast convolution algorithms. In this paper we shortly sum up some of the common approaches and present a straightforward method for bandlimited discrete-time realization of analog nonlinear audio effects, like tube amps, exciters etc., using off-time digital cross correlation measurements. From these measurements we obtain a rather inefficient Wiener representation of the unknown nonlinearity. We then reduce the number of required coefficients significantly on the basis of multi-dimensional Laguerre transformation of the related Volterra kernels to allow real-time implementation on a digital signal processor [5].
Download Audio-rate control of FFT based processing using few parameters Though the use of the Fast Fourier Transform (FFT) for signal processing in music applications has been widespread, applications in real-time systems for dynamic spectral transformation has been quite limited. The limitations have been largely due to amount of computation required for the operations. With faster machines, and with suitable implementations for frequency-domain processing, real-time dynamic control of high-quality spectral processing can be accomplished with great efficiency and simple approach. This paper will focus on dynamic real-time control of frequencydomain-based signal processing, and will describe the author's latest work (hi-resolution filtering and spatialization implementations) in this area. General background on the implementation and the development environment (Max Signal Processing, MSP) will also be provided.
Download Effect of early reflections in binaural systems with loudspeaker reproduction Systems for 3D sound reproduction are often implemented with binaural technology where signals are played back over loudspeakers. This paper reports preliminary results from an investigation on how reflected sound in the listening room influences horisontal localisation in such systems. An experiment, consisting of listening tests, was done. Results from the experiment showed that reflections as late as 5ms and 10ms did influence localisation in such systems. The probability for reversals between front and back localisation increased, and the ability to localise to the back was degraded. Localisation was clustered towards the direction of the reflections.
Download CA: A system for Granular Processing of Sound using Cellular Automata is a tool for the granular processing of sound using cellular automata developed on the SGI-Indy platform. It investigates the effects of change in the timbre of sound using a cellular automaton in real-time. The cellular automaton generated by the chosen rule controls parameters of a bank of filters. The system uses standard infinite impulse response filters and a general model of three neighborhood cellular automata. The composer1 can configure the filter banks by adjusting bandwidths and center frequencies through the graphical interface. CA is very well suited as a tool for computer music composition because it is capable of creating a new palette of sounds for the composer and it is easy to use.
Download Identification and Modeling of a Flute Source Signal This paper addresses the modeling of the source signal of a flute sound obtained by «removing» the contribution of the resonator. The resulting sound has then a more regular spectral behavior and can be modeled using signal models. The decomposition of the source signal into a deterministic and a stochastic part has been made using adaptive filtering. The deterministic part can then be modeled by non-linear synthesis models, the parameters of which are obtained using perceptive criteria. Linear filtering are used to model the stochastic part of the source signal.
Download Digital sound synthesis, acoustics and perception: a rich intersection The early years of digital sound synthesis were filled with promise following Max Mathews’ publication in 1963 of his pioneering work at Bell Telephone Laboratories [1]. The digital control of loudspeakers allowed for the production of any conceivable sound given the correct sequence of numbers (samples). Producing the correct sequence of numbers, however, turned out to be a formidable task. Acoustics and psychoacoustics, the first a well-developed field of knowledge and the second less so, did not provide information at the level of detail required to simulate even the simplest sound of an acoustic instrument. The enormous potential of digital synthesis counterpoised with an enormous knowledge deficit were the initial conditions for interdisciplinary research that continues to this day. Discoveries have been made and insights gained that are of consequence in the general field of digital audio.
Download Real-time time-varying frequency warping via short-time Laguerre transform In this paper we address the problem of the real-time implementation of time-varying frequency warping. Frequency warping based on a one-parameter family of one-to-one warping maps can be realized by means of the Laguerre transform and implemented in a non-causal structure. This structure is not directly suited for real-time implementation since each output sample is formed by combining all of the input samples. Similarly, the recently proposed time-varying Laguerre transform has the same drawback. Furthermore, long frequency dependent delays destroy the time organization or macrostructure of the sound event. Recently, the author has introduced the Short-Time Laguerre Transform for the approximate real-time implementation of frequency warping. In this transform the short-time spectrum rather than the overall frequency spectrum is frequency warped. The input is subdivided into frames that are tapered by a suitably selected window. By careful design, the output frames correspond to warped versions of the input frames modulated by a stretched version of the window. It is then possible to overlap-add these frames without introducing audible distortion. The overlap-add technique can be generalized to time-varying warping. However, several issues concerning the design of the window and the selection of the overlap parameters need to be addressed. In this paper we discuss solutions for the overlap of the frames when the Laguerre parameter is kept constant but distinct in each frame and solutions for the computation of full time-varying frequency warping when the Laguerre parameter is changing within each frame.
Download A virtual DSP architecture for MPEG-4 structured audio The MPEG-4 Audio standard provides a toolset for synthetic Audio generation and Audio processing called Structured Audio (SA). SA permits to describe algorithms through its Structured Audio Orchestra Language (SAOL) programming language. Unlike some other languages of the same type, SAOL has a sample-by-sample execution structure, and this makes particularly important the overhead computation in the case of an interpreted decoding. This paper describes the design of a virtual DSP architecture able to exploit the data level parallelism contained in many audio synthesis and processing algorithms and to consistently reduce the implementation overhead.
Download Efficient linear prediction for digital audio effects In many audio applications an appropriate spectral estimation from a signal sequence is required. A common approach for this task is the linear prediction [1] where the signal spectrum is modelled by an all-pole (purely recursive) IIR (infinite impulse response) filter. Linear prediction is commonly used for coding of audio signals leading to linear predictive coding (LPC). But also some audio effects can be created using the spectral estimation of LPC. In this paper we consider the use of LPC in a real-time system. We investigate several methods of calculating the prediction coefficients to have an almost fixed workload each sample. We present modifications of the autocorrelation method and of the Burg algorithm for a sample-based calculation of the filter coefficients as alternative for the gradient adaptive lattice (GAL) method. We discuss the obtained prediction gain when using these methods regarding the required complexity each sample. The desired constant workload leads to a fast update of the spectral model which is of great benefit for both coding and audio effects.
Download Interactive digital audio environments: gesture as a musical parameter This paper presents some possible relationships between gesture and sound that may be built with an interactive digital audio environment. In a traditional musical situation gesture usually produces sound. The relationship between gesture and sound is unique, it is a cause to effect link. In computer music, the possibility of uncoupling gesture from sound is due to the fact that computer can carry out all the aspects of sound production from composition up to interpretation and performance. Real time computing technology and development of human gesture tracking systems may enable gesture to be introduced again into the practice of computer music, but with a completely renewed approach. There is no more need to create direct cause to effect relationships for sound production, and gesture may be seen as another musical parameter to play with in the context of interactive musical performances.