In this paper, we propose a new feature selection algorithm for multi-class classification. The proposed algorithm is based on Gaussian mixture models (GMMs) of the features, and it uses the distance between the two least separable classes as a metric for feature selection. The proposed system was tested with a support vector machine (SVM) for multi-class classification of music. Results show that the proposed feature selection scheme is superior to conventional schemes.

In this paper, we propose a robust room inverse filtering algorithm for speech dereverberation based on a kurtosis maximization. The proposed algorithm utilizes a new normalized kurtosis function that nonlinearly maps the input kurtosis onto a finite range from zero to one, which results in a kurtosis warping. Due to the kurtosis warping, the proposed algorithm provides more stable convergence and, in turn, better performance than the conventional algorithm. Experimental results are presented to confirm the robustness of the proposed algorithm.

This paper proposes an efficient FFT algorithm for the Psycho-Acoustic Model (PAM) of MPEG-4 AAC. The proposed algorithm synthesizes FFT coefficients using MDCT and MDST coefficients through circular convolution. The complexity of the MDCT and MDST coefficients is approximately half of the original FFT. We also design a new PAM based on the proposed FFT algorithm, which has 15% lower computational complexity than the original PAM without degradation of sound quality. Subjective as well as objective test results are presented to confirm the efficiency of the proposed FFT computation algorithm and the PAM.

We present a state-space approach to the problem of designing a parallel feedforward compensator (PFC), which has the same dimension of the input i.e. input-dimensional, for a class of non-square linear systems such that the closed-loop system is strictly passive. For a non-minimum phase system or a system with high relative degree, passification of the system cannot be achieved by any other methodologies except by using a PFC. In our scheme, we first determine a squaring gain matrix and an additional dynamics that is connected to the system in a feedforward way, then a static passifying control law is designed. Consequently, the actual feedback controller will be the static control law combined with the feedforward dynamics. Necessary and sufficient conditions for the existence of the PFC are given by the static output feedback formulation, which enables to utilize linear matrix inequality (LMI). Since the proposed PFC is input-dimensional, our design procedure can be viewed as a solution to the low-order dynamic output feedback control problem in the literature. The effectiveness of the proposed method is illustrated by some numerical examples.

This paper presents an approximated virtual source imaging system based on crosstalk cancellation with a pair of closely spaced loudspeakers. Utilizing the frequency-dependent relative importance of sound localization cues, the proposed system provides separate approximations for the low- and high-frequency bands. Experimental results show that the system provides good approximations within ±55° in the stereo dipole setup with natural sound quality.

This paper proposes a method of improving the performance of blind reverberation time (RT) estimation in noisy environments. RT estimation is conducted using a maximum likelihood (ML) method based on the autocorrelation function of the linear predictive residual signal. To reduce the effect of environmental noise, a noise reduction technique is applied to the noisy speech signal. In addition, a frequency coefficient selection is performed to eliminate signal components with low signal-to-noise ratio (SNR). Experimental results confirm that the proposed method improves the accuracy of RT measures, particularly when the speech signal is corrupted by a colored noise with a narrow bandwidth.

In this paper, we propose a switchable linear prediction (LP)/warped linear prediction (WLP) hybrid scheme for the transform coded excitation (TCX) coder, which is adopted as a core codec in AMR-WB+ and USAC. The proposed algorithm selects either an LP or WLP filter on a per-frame basis. To provide a smooth transitions between LP and WLP frames, a window switching scheme is developed using sine and rectangular windows. In addition, a Gaussian Mixture Model (GMM)-based classification module is used to determine the prediction mode. Through a subjective listening test it was confirmed that the proposed LP/WLP switching scheme offers improved sound quality.

In this letter, we propose an acoustic distance rendering (ADR) algorithm that can efficiently create the proximity effect in virtual reality (VR) systems. By observing the variation of acoustic cues caused by the movement of the sound source in the near field, we develop a model that can closely approximates the near-field transfer function (NFTF). The developed model is used to efficiently compensate for the near-field effect on the head related transfer function (HRTF). The proposed algorithm is implemented and tested in the form of an audio plugin for a VR platform and the test results confirm the efficiency of the proposed algorithm.

An efficient active noise control algorithm based on the lattice-transversal joint (LTJ) filter structure is presented, and applied to the active control of broadband noise in a 3-dimensional enclosure. The presented algorithm implements the filtered-x LMS within the LTJ structure obtained by cascading the lattice and transversal structures. Simulation results show that the LTJ-based noise control algorithm has fast convergence speed that is comparable to the lattice-based algorithm while its computational complexity is less demanding.

Development of an artificial reverberator for low-memory requirements is an issue of importance in applications such as mobile multimedia devices. One possibility is to use an All-Pass Filter (APF), which is embedded in the feedback loop of the comb filter network. In this paper, we propose a reverberator employing time-varying APFs to increase the reverberation performance. By changing the gain of the APF, we can increase the number of frequency peaks perceptually. Thus, the resulting reverberation sounds much more natural, even with less memory, than the conventional approach. In this paper, we perform theoretical and perceptual analyses of artificial reverberators employing time-varying APF. Through the analyses, we derive the degree of phase variation of the APF that is perceptually acceptable. Based on the analyses, we propose a method of designing artificial reverberators associated with the time-varying APFs. Through subjective tests, it is shown that the proposed method is capable of providing perceptually comparable sound quality to the conventional methods even though it uses less memory.

Although the AMR-WB+ coder provides excellent quality for speech signal, its coding model for music signals is not as optimal as the HE-AAC v2. The main causes of the poor quality of the AMR-WB+ TCX are the non-critical sampling and block artifacts. The new TCX windowing scheme proposed in this paper uses an MDCT with a 50% frame overlap, so that the problems of non-critical sampling and blocking artifacts are significantly mitigated. Due to long overlaps, the proposed scheme involves an additional codec delay. It is, however, moderate for audio services. The results of objective and subjective tests indicate that the proposed scheme achieves noticeable quality improvements for music signals over the previous TCX schemes.