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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