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.
Download Digital guitar body mode modulation with one driving parameter
In this study we have developed a digital guitar body mode modulation technique where the modulation can be controlled through one driving parameter. The filtering and modulation is done with frequency-warped recursive filters that have been implemented in real-time on a modern DSP processor. By changing the warping parameter the perceived size of the body can be controlled, by a pedal or automatically, resulting in an interesting effect. This effect is useful both for the electric and the amplified acoustic guitar. Perceptual properties of the effect are studied by a listening experiment. (See also
Download Traditional (?) implementations of a phase vocoder: the tricks of the trade
Download Model-based synthesis and transformation of voiced sounds
In this work a glottal model loosely based on the Ishizaka and Flanagan model is proposed, where the number of parameters is drastically reduced. First, the glottal excitation waveform is estimated, together with the vocal tract filter parameters, using inverse filtering techniques. Then the estimated waveform is used in order to identify the nonlinear glottal model, represented by a closedloop configuration of two blocks: a second order resonant filter, tuned with respect to the signal pitch, and a regressor-based functional, whose coefficients are estimated via nonlinear identification techniques. The results show that an accurate identification of real data can be achieved with less than regressors of the nonlinear functional, and that an intuitive control of fundamental features, such as pitch and intensity, is allowed by acting on the physically informed parameters of the model. 10
Download Digital waveguide networks as multidimensional wave digital filters
Download An auditorily motivated analysis method for room impulse responses
In this paper a new auditorily motivated analysis method for room impulse responses is presented. The method applies same kind of time and frequency resolution than the human hearing. With the proposed method it is possible to study the decaying sound field of a room in more detail. It is applicable as well in the analysis of artificial reverberation and related audio effects. The method, used with directional microphones, gives us also hints about the diffuseness and the directional characteristics of the sound fields in the time-frequency domain. As a case study two example room impulse responses are analyzed.