A minimax approach for the joint design of acoustic crosstalk cancellation filters

Journal ArticleAbstract-This paper presents a method for jointly designing immersive audio rendering filters for a single listener using loudspeakers. The filters for crosstalk cancellation are assumed to have finite impulse responses and are designed using the minimax criterion. In addition to the traditional Atal-Schroeder crosstalk canceler structure, this paper explores an alternate topology that requires the approximation of a single filter. In general, the minimax approach provides improved low-frequency performance leading to a better overall separation of the direct-path and cross-path transfer functions than least-squares designs. The performance of the single-filter structure is better than that of the traditional crosstalk cancellation structure

Design of FIR digital filters for pulse shaping and channel equalization using time-domain optimization

Three algorithms are developed for designing finite impulse response digital filters to be used for pulse shaping and channel equalization. The first is the Minimax algorithm which uses linear programming to design a frequency-sampling filter with a pulse shape that approximates the specification in a minimax sense. Design examples are included which accurately approximate a specified impulse response with a maximum error of 0.03 using only six resonators. The second algorithm is an extension of the Minimax algorithm to design preset equalizers for channels with known impulse responses. Both transversal and frequency-sampling equalizer structures are designed to produce a minimax approximation of a specified channel output waveform. Examples of these designs are compared as to the accuracy of the approximation, the resultant intersymbol interference (ISI), and the required transmitted energy. While the transversal designs are slightly more accurate, the frequency-sampling designs using six resonators have smaller ISI and energy values

Optimization-based reproduction of diffuse audio objects

The creation of a diffuse sound event from a single audio signal is an important signal processing task, for instance in spatial audio reproduction or audio coding. Current algorithms based on decorrelation filters or frequency-dependent panning typically cause artifacts due to transients or time-domain aliasing. In this paper, we propose an optimization-based approach to diffusion that creates a set of filters to approximate a desired distribution of frequency-dependent propagation directions to create the perception of a diffuse sound field with a multi-channel audio system. Thus, the diffusion can be optimally adapted to a specific reproduction scenario. In addition, the transient response can be purposefully improved by imposing constraints on the time-domain filter coefficients

On the indoor beamformer design with reverberation

Beamforming remains to be an important technique for signal enhancement. For applications in open space, the transfer function describing waves propagation has an explicit expression, which can be employed for beamformer design. However, the function becomes very complex in an indoor environment due to the effects of reverberation. In this paper, this problem is discussed. A method based on the image source method (ISM) is applied to model the room impulse responses (RIRs), which will act as the transfer function between source and sensor. The indoor beamformer design problem is formulated as a minimax optimization problem. We propose and study several optimization models based on the -norm to design the beamformer. We found that it is advantageous to separate early and late reverberations in the design process and better designs can be achieved. Several numerical experiments are presented using both simulated data and real recordings to evaluate the proposed methods