Download A Virtual Tube Delay Effect A virtual tube delay effect based on the real-time simulation of acoustic wave propagation in a garden hose is presented. The paper describes the acoustic measurements conducted and the analysis of the sound propagation in long narrow tubes. The obtained impulse responses are used to design delay lines and digital filters, which simulate the propagation delay, losses, and reflections from the end of the tube which may be open, closed, or acoustically attenuated. A study on the reflection caused by a finite-length tube is described. The resulting system consists of a digital waveguide model and produces delay effects having a realistic low-pass filtering. A stereo delay effect plugin in P URE DATA1 has been implemented and it is described here.
Download Analysis/Synthesis of the Andean Quena via Harmonic Band Wavelet Transform It is well known that one of the challenges in musical instruments analysis is to obtain relevant signal characteristics and information for sound description and classification. In this paper we study the Peruvian quena flute by means of the Harmonic Band Wavelet Transform (HBWT), a convenient representation for the sound content based on its 1/f fractal characteristics. In order to identify a relationship between fractal characteristics of musical sounds, we developed two sound transformations and establish a comparison between quena, a recorder and melodica wind instruments. The sound transformations implemented were noise filtering and pitch-shifting while the sound classification was focused on the γp fractal attribute. Our work led us to the conclusion that the HBWT quena representation favored the implementation of sound transformations and that the γp fractal feature had great potential in musical instruments recognition and classification applications. Keywords: sound fractal analysis; quena; 1/f noise; noise filtering; pitch-shifting.
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 Perceptual Audio Source Culling for Virtual Environments Existing game engines and virtual reality software, use various techniques to render spatial audio. One such technique, binaural synthesis, is achieved through the use of head-related transfer functions, in conjunction with artificial reverberators. For virtual environments that embody a large number of concurrent sound sources, binaural synthesis will be computationally costly. The work presented in this paper aims to develop a methodology that improves overall performance by culling inaudible and perceptually less prominent sound sources in order to reduce performance implications. The proposed algorithm is benchmarked and compared with distance-based, volumetric culling methodology. A subjective evaluation of the perceptual performance of the proposed algorithm for acoustic scenes having different compositions is also provided.
Download Binauralization of Omnidirectional Room Impulse Responses - Algorithm and Technical Evaluation The auralization of acoustic environments over headphones is often realized with data-based dynamic binaural synthesis. The required binaural room impulse responses (BRIRs) for the convolution process can be acquired by performing measurements with an artificial head for different head orientations and positions. This procedure is rather costly and therefore not always feasible in practice. Because a plausible representation is sufficient for many practical applications, a simpler approach is of interest. In this paper we present the BinRIR (Binauralization of omnidirectional room impulse responses) algorithm, which synthesizes BRIR datasets for dynamic auralization based on a single measured omnidirectional room impulse response (RIR). Direct sound, early reflections, and diffuse reverberation are extracted from the omnidirectional RIR and are separately spatialized. Spatial information is added according to assumptions about the room geometry and on typical properties of diffuse reverberation. The early part of the RIR is described by a parametric model and can easily be modified and adapted. Thus the approach can even be enhanced by considering modifications of the listener position. The late reverberation part is synthesized using binaural noise, which is adapted to the energy decay curve of the measured RIR. In order to examine differences between measured and synthesized BRIRs, we performed a technical evaluation for two rooms. Measured BRIRs are compared to synthesized BRIRs and thus we analyzed the inaccuracies of the proposed algorithm.
Download Neural Modelling of Time-Varying Effects This paper proposes a grey-box neural network based approach
to modelling LFO modulated time-varying effects.
The neural
network model receives both the unprocessed audio, as well as
the LFO signal, as input. This allows complete control over the
model’s LFO frequency and shape. The neural networks are trained
using guitar audio, which has to be processed by the target effect
and also annotated with the predicted LFO signal before training.
A measurement signal based on regularly spaced chirps was used
to accurately predict the LFO signal. The model architecture has
been previously shown to be capable of running in real-time on a
modern desktop computer, whilst using relatively little processing
power. We validate our approach creating models of both a phaser
and a flanger effects pedal, and theoretically it can be applied to
any LFO modulated time-varying effect. In the best case, an errorto-signal ratio of 1.3% is achieved when modelling a flanger pedal,
and previous work has shown that this corresponds to the model
being nearly indistinguishable from the target device.
Download A Wave Digital Filter Model of the Fairchild 670 Limiter This paper presents a circuit-based, digital model of the prized 1950’s vintage Fairchild R 670 vacuum tube limiter. The model uses a mixture of black boxes and wave digital filters, as a step toward a fully wave digital filter design. Wave digital filters provide an efficient, modular way to digitally simulate analog circuits. A novel model for the 6386 triode is introduced to simulate the active component in a wave digital filter model of the Fairchild 670’s signal amplifier. The signal amplifier is integrated with a hybrid wave digital filter/black-box sidechain amplifier model to form a complete model of the Fairchild 670. Model test results for music and pure tones are discussed, highlighting the device’s static gain characteristics and gain reduction dependent distortion. Finally, this paper discusses the model’s salient features and their implications for designing dynamics processors.
Download Model Bending: Teaching Circuit Models New Tricks A technique is introduced for generating novel signal processing systems grounded in analog electronic circuits, called model bending. By applying the ideas behind circuit bending to models of nonlinear analog circuits it is possible to create novel nonlinear signal processors which mimic the behavior of analog electronics, but which are not possible to implement in the analog realm. The history of both circuit bending and circuit modeling is discussed, as well as a theoretical basis for how these approaches can complement each other. Potential pitfalls to the practical application of model bending are highlighted and suggested solutions to those problems are provided, with examples.
Download Stable Limit Cycles as Tunable Signal Sources This paper presents a method for synthesizing audio signals from
nonlinear dynamical systems exhibiting stable limit cycles, with
control over frequency and amplitude independent of changes to
the system’s internal parameters. Using the van der Pol oscillator
and the Brusselator as case studies, it is demonstrated how parameters are decoupled from frequency and amplitude by rescaling the
angular frequency and normalizing amplitude extrema. Practical
implementation considerations are discussed, as are the limits and
challenges of this approach. The method’s validity is evaluated experimentally and synthesis examples show the application of tunable nonlinear oscillators in sound design, including the generation
of transients in FM synthesis by means of a van der Pol oscillator
and a Supersaw oscillator bank based on the Brusselator.
Download Additive Synthesis Of Sound By Taking Advantage Of Psychoacoustics In this paper we present an original technique designed in order to speed up additive synthesis. This technique consists in taking into account psychoacoustic phenomena (thresholds of hearing and masking) in order to ignore the inaudible partials during the synthesis process, thus saving a lot of computation time. Our algorithm relies on a specific data structure called “skip list” and has proven to be very efficient in practice. As a consequence, we are now able to synthesize an impressive number of spectral sounds in real time, without overloading the processor.