This paper presents an automated Public Address processing unit, using delay and magnitude response adjustment. The aim is to achieve a flat frequency response and delay adjustment between different physically-placed speakers at the measuring point, which is nowadays usually made manually by the sound technician. The adjustment is obtained using three signal processing operations to the audio signal: time delay adjustment, crossover filtering, and graphic equalization. The automation is in the calculation of different parameter sets: estimation of the time delay, the selection of a suitable crossover frequency, and calculation of the gains for a third-octave graphic equalizer. These automatic methods reduce time and effort in the calibration of line-array PA systems, since only three sine sweeps must be played through the sound system. Measurements have been conducted in an anechoic chamber using a 1:10 scale model of a line array system to verify the functioning of the automatic calibration and equalization methods.

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This paper describes the implementation of a recently introduced parallel graphic equalizer (PGE) in a heterogeneous way. The control and audio signal processing parts of the PGE are distributed to a PC and to a signal processor, of WaveCore architecture, respectively. This arrangement is particularly suited to the algorithm in question, benefiting from the low-latency characteristics of the audio signal processor as well as general purpose computing power for the more demanding filter coefficient computation. The design is achieved cleanly in a high-level language called Kronos, which we have adapted for the purposes of heterogeneous code generation from a uniform program source.

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A method to measure the response of a linear time-variant (LTV) audio system is presented. The proposed method uses a series of short chirps generated as the impulse response of several cascaded allpass filters. This test signal can measure the characteristics of an LTV system as a function of time. Results obtained from testing of this method on a guitar phaser pedal are presented. A proof of concept gray-box model of the measured system is produced based on partial knowledge about the internal structure of the pedal and on the spectral analysis of the measured responses. The temporal behavior of the digital model is shown to be very similar to that of the measured device. This demonstrates that it is possible to measure LTV analog audio systems and produce approximate virtual analog models based on these results.

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The use of the bandlimited ramp (BLAMP) function as an antialiasing tool for audio signals with sharp corners is presented. Discontinuities in the waveform of a signal or its derivatives require infinite bandwidth and are major sources of aliasing in the digital domain. A polynomial correction function is modeled after the ideal BLAMP function. This correction function can be used to treat aliasing caused by sharp edges or corners which translate into discontinuities in the first derivative of a signal. Four examples of cases where these discontinuities appear are discussed: synthesis of triangular waveforms, hard clipping, and half-wave and fullwave rectification. Results obtained show that the BLAMP function is a more efficient tool for alias reduction than oversampling. The polynomial BLAMP can reduce the level of aliasing components by up to 50 dB and improve the overall signal-to-noise ratio by about 20 dB. The proposed method can be incorporated into virtual analog models of musical systems.

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The design of graphic equalizers has been investigated for decades, but only recently fitting the magnitude response closely enough to the control points has become possible. This paper reviews the development of graphic equalizer design and discusses how to define the target response. Furthermore, it investigates how to find the hardest target gain settings, the definition of the bandwidth of band filters, the estimation of the interaction between the bands, and how the number of iterations improves the design. The main focus is on a recent design principle for the cascade graphic equalizer. This paper extends the design method for the case of third-octave bands, showing how to choose the parameters to obtain good accuracy. The main advantages of the proposed approach are that it keeps the approximation error below 1 dB using only a single second-order IIR filter section per band, and that its design is fast. The remaining challenge is to simplify the design phase so that sufficient accuracy can be obtained without iterations.

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An antialiased digital model of the wavefolding circuit inside the Buchla 259 Complex Waveform Generator is presented. Wavefolding is a type of nonlinear waveshaping used to generate complex harmonically-rich sounds from simple periodic waveforms. Unlike other analog wavefolder designs, Buchla’s design features five op-amp-based folding stages arranged in parallel alongside a direct signal path. The nonlinear behavior of the system is accurately modeled in the digital domain using memoryless mappings of the input–output voltage relationships inside the circuit. We pay special attention to suppressing the aliasing introduced by the nonlinear frequency-expanding behavior of the wavefolder. For this, we propose using the bandlimited ramp (BLAMP) method with eight times oversampling. Results obtained are validated against SPICE simulations and a highly oversampled digital model. The proposed virtual analog wavefolder retains the salient features of the original circuit and is applicable to digital sound synthesis.

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Decorrelation of audio signals is an important process in the spatial reproduction of sounds. For instance, a mono signal that is spread on multiple loudspeakers should be decorrelated for each channel to avoid undesirable comb-filtering artifacts. The process of decorrelating the signal itself is a compromise aiming to reduce the correlation as much as possible while minimizing both the sound coloration and the computing cost. A popular decorrelation method, convolving a sound signal with a short sequence of exponentially decaying white noise which, however, requires the use of the FFT for fast convolution and may cause some latency. Here we propose a decorrelator based on a sparse random sequence called velvet noise, which achieves comparable results without latency and at a smaller computing cost. A segmented temporal decay envelope can also be implemented for further optimizations. Using the proposed method, we found that a decorrelation filter, of similar perceptual attributes to white noise, could be implemented using 87% less operations. Informal listening tests suggest that the resulting decorrelation filter performs comparably to an equivalent white-noise filter.

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This paper proposes signal processing methods to extend a stationary part of an audio signal endlessly. A frequent occasion is that there is not enough audio material to build a synthesizer, but an example sound must be extended or modified for more variability. Filtering of a white noise signal with a filter designed based on high-order linear prediction or concatenation of the example signal can produce convincing arbitrarily long sounds, such as ambient noise or musical tones, and can be interpreted as a spectral freeze technique without looping. It is shown that the random input signal will pump energy to the narrow resonances of the filter so that lively and realistic variations in the sound are generated. For realtime implementation, this paper proposes to replace white noise with velvet noise, as this reduces the number of operations by 90% or more, with respect to standard convolution, without affecting the sound quality, or by FFT convolution, which can be simplified to the randomization of spectral phase and only taking the inverse FFT. Examples of producing endless airplane cabin noise and piano tones based on a short example recording are studied. The proposed methods lead to a new way to generate audio material for music, films, and gaming.

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Decorrelation of audio signals is a critical step for spatial sound reproduction on multichannel configurations. Correlated signals yield a focused phantom source between the reproduction loudspeakers and may produce undesirable comb-filtering artifacts when the signal reaches the listener with small phase differences. Decorrelation techniques reduce such artifacts and extend the spatial auditory image by randomizing the phase of a signal while minimizing the spectral coloration. This paper proposes a method to optimize the decorrelation properties of a sparse noise sequence, called velvet noise, to generate short sparse FIR decorrelation filters. The sparsity allows a highly efficient time-domain convolution. The listening test results demonstrate that the proposed optimization method can yield effective and colorless decorrelation filters. In comparison to a white noise sequence, the filters obtained using the proposed method preserve better the spectrum of a signal and produce good quality broadband decorrelation while using 76% fewer operations for the convolution. Satisfactory results can be achieved with an even lower impulse density which decreases the computational cost by 88%.

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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.

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