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 High-level musical control paradigms for Digital Signal Processing No matter how complex DSP algorithms are and how rich sonic processes they produce, the issue of their control immediately arises when they are used by musicians, independently on their knowledge of the underlying mathematics or their degree of familiarity with the design of digital instruments. This text will analyze the problem of the control of DSP modules from a compositional standpoint. An implementation of some paradigms in a Lisp-based environment (omChroma) will also be concisely discussed.
Download Continuous and discrete Fourier spectra of aperiodic sequences for sound modeling The Fourier analysis of aperiodic ordered time structures related with number eight is considered. Recursion relations for the Fourier amplitudes are obtained for a sequence with discrete spectrum. The continuous spectrum of a different type of sequence is also studied . By increasing the number of points in the time axis dynamic spectra can be obtained and used for sound synthesis.
Download Monophonic transcription with autocorrelation This paper describes an algorithm, which performs monophonic music transcription. A pitch tracker calculates the fundamental frequency of the signal from the autocorrelation function. A continuity-restoration block takes the extracted pitch and determines the score corresponding to the original performance. The signal envelope analysis completes the transcription system, calculating attack-sustain-decay-release times, which improves the synthesis process. Attention is also paid to the extraction of timbre and wavetable synthesis.
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.
Download A reverberator based on absorbent all-pass filters Artificial reverberator topologies making use of all-pass filters in a feedback loop are popular, but have lacked accurate control of decay time and energy level. This paper reviews a general theory of artificial reverberators based on Unitary-Feedback Delay Networks (UFDN), which allow accurate control of the decay time at multiple frequencies in such topologies. We describe the design of an efficient reverberator making use of chains of elementary filters, called “absorbent all-pass filters”, in a feedback loop. We show how, in this particular topology, the late reverberant energy level can be controlled independently of the other control parameters. This reverberator uses the I3DL2 control parameters, which have been designed as a standard interface for controlling reverberators in interactive 3D audio.
Download Visualization and calculation of the roughness of acoustical musical signals using the Synchronization Index Model (SIM) The synchronization index model of sensory dissonance and roughness accounts for the degree of phase-locking to a particular frequency that is present in the neural patterns. Sensory dissonance (roughness) is defined as the energy of the relevant beating frequencies in the auditory channels with respect to the total energy. The model takes rate-code patterns at the level of the auditory nerve as input and outputs a sensory dissonance (roughness) value. The synchronization index model entails a straightforward visualization of the principles underlying sensory dissonance and roughness, in particular in terms of (i) roughness contributions with respect to cochlear mechanical filtering (on a Critical Band scale), and (ii) roughness contributions with respect to phase-locking synchrony (=the synchronization index for the relevant beating frequencies on a frequency scale). This paper presents the concept, and implementation of the synchronization index model and its application to musical scales.
Download A system for data-driven concatenative sound synthesis In speech synthesis, concatenative data-driven synthesis methods prevail. They use a database of recorded speech and a unit selection algorithm that selects the segments that match best the utterance to be synthesized. Transferring these ideas to musical sound synthesis allows a new method of high quality sound synthesis. Usual synthesis methods are based on a model of the sound signal. It is very difficult to build a model that would preserve the entire fine details of sound. Concatenative synthesis achieves this by using actual recordings. This data-driven approach (as opposed to a rule-based approach) takes advantage of the information contained in the many sound recordings. For example, very naturally sounding transitions can be synthesized, since unit selection is aware of the context of the database units. The C ATERPILLAR software system has been developed to allow data-driven concatenative unit selection sound synthesis. It allows high-quality instrument synthesis with high level control, explorative free synthesis from arbitrary sound databases, or resynthesis of a recording with sounds from the database. It is based on the new software-engineering concept of component-oriented software, increasing flexibility and facilitating reuse.
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.