Whereas the autocorrelation is frequently used for pitch estimation, the resultant estimates usually suffer from inaccuracy. Instead of upsampling, we can improve the accuracy of the estimates by applying polynomial interpolation to the autocorrelation directly. For that purpose, four kernels, which are interpolating quadratic, quadratic-B spline, cubic-B spline, and cubic convolution kernels respectively, have been compared. Experiments show that the cubic B spline kernel shows the best performance, a little inferior to the computationally intensive upsampling procedure. The quadratic B spline kernel shows also reasonable performance with the merit of the further reduced computational complexities compared with the cubic B spline kernel.

We analyze the effect of window choice on the zero-padding method and corrected quadratically interpolated fast Fourier transform using a harmonic signal in noise at both high and low signal-to-noise ratios (SNRs) on a theoretical basis. Then, we validate the theoretical analysis using simulations. The theoretical analysis and simulation results using four traditional window functions show that the optimal window is determined depending on the SNR; the estimation errors are the smallest for the rectangular window at low SNR, the Hamming and Hanning windows at mid SNR, and the Blackman window at high SNR. In addition, we analyze the simulation results using the signal-to-noise floor ratio, which appears to be more effective than the conventional SNR in determining the optimal window.

We propose a new context-adaptive arithmetic coding (CAAC) scheme for lossless bit rate reduction of parametric stereo (PS) in enhanced aacPlus. Based on the probability analysis of stereo parameters indexes in PS, we propose a stereo band-dependent CAAC scheme for PS. We also propose a new coding structure of the scheme which is simple but effective. The proposed scheme has normal and memory-reduced versions, which are superior to the original and conventional schemes and guarantees significant bit rate reduction of PS. The proposed scheme can be an alternative to the original PS coding scheme at low bit rate, where coding efficiency is very important.

Recently, array speaker products have received attention in the field of consumer electronics, and control technologies for arrayed speaker units, including beamforming and wave field synthesis (WFS), have been developed for various purposes. An important application of these algorithms is focused source reproduction. The focused source reproduction capability is strongly coupled with the array length. The array length is a very important design factor in consumer products, but it is very short in home entertainment systems, compared with ideal WFS systems or theater speaker systems. Therefore, a well-defined measure for the maximum focusing range is necessary for designing an array speaker product. In this paper, a maximum focusable range measure is proposed and is analyzed by simulation of a small array speaker. The analysis results show that the proposed maximum focusable range has properties strongly related to the capability for focused source reproduction.

We propose a new coding scheme for lossless bit rate reduction of the MPEG Surround module in unified speech and audio coding (USAC). The proposed scheme is based on context-adaptive arithmetic coding for efficient bit stream composition of spatial parameters. Experiments show that it achieves the significant lossless bit reduction of 9.93% to 12.14% for spatial parameters and 8.64% to 8.96% for the overall MPEG Surround bit streams compared to the original scheme. The proposed scheme, which is not currently included in USAC, can be used for the improved coding efficiency of MPEG Surround in USAC, where the saved bits can be utilized by the other modules in USAC.

Recent studies show that several FFT-based high-accuracy frequency estimation methods achieve very good performance. In this letter, we select three methods, which are the zero-padding, weighted multipoint interpolated DFT, and phase difference approximation respectively, and discuss the window selection for each method. Experiments show that the window selection primarily depends on the signal-to-noise ratio (SNR). As a result, the optimal window selection for each method and, reversely, the optimal selection of the estimation method for a specific window are discussed as a function of SNR. Consideration on the computational load and the resolution problem is also briefly discussed.

An efficient method is proposed for reconstructing speakerphone-mode cellular phone sound. The overall transfer function from digital PCM signals stored in a cellular phone to dummy head-recorded signals is modeled as a combination of a cellular phone transfer function (CPTF) and a cellular phone-to-listener transfer function (CPLTF). The CPTF represents the linear and nonlinear characteristics of a cellular phone and is modeled by the Volterra model. The CPLTF represents the effect of the path from a cellular phone to a dummy head and is measured. Listening tests show the effectiveness of the proposed method. An application scenario of the proposed method is also addressed for sound quality assessment of cellular phones in speakerphone mode.

The development of multichannel audio systems has increased the need for multichannel contents. However, the supply of multichannel audio contents is not sufficient for advanced multichannel systems. Therefore, home entertainment manufacturers need upmixing systems, including systems that utilize monaural time-frequency domain information. Therefore, a monaural ambience extraction algorithm based on nonnegative matrix factorization (NMF) has been developed recently. Ambience signals refer to sound components that do not have obvious spatial images, e.g., wind, rain, and diffuse sound. The developed algorithm provides good upmixing performance; however, the algorithm is a batch process and therefore, it cannot be used by home audio manufacturers. In this paper, we propose an on-line monaural ambience extraction algorithm. The proposed algorithm analyzes the dominant components with an on-line NMF algorithm, and extracts the remaining sound as ambience components. Experiments were performed with artificial mixed signals and real music signals, and the performance of the proposed algorithm was compared with the performance of the conventional batch algorithm as a reference. The experimental results show that the proposed algorithm extracts the ambience components as well as the batch algorithm, despite the on-line constraints.

We propose an iterative frequency estimation method for accuracy improvement of discrete Fourier transform (DFT) phase-based methods. It iterates frequency estimation and phase calculation based on the DFT phase-based methods, which maximizes the signal-to-noise floor ratio at the frequency estimation position. We apply it to three methods, the phase difference estimation, the derivative estimation, and the arctan estimation, which are known to be among the best DFT phase-based methods. Experimental results show that the proposed method shows meaningful reductions of the frequency estimation error compared to the conventional methods especially at low signal-to-noise ratio.

A simple but effective fundamental frequency estimation method is proposed using parametric cubic convolution. The performance of the method is shown to be good not only for the stationary signals but also for the signal whose fundamental frequency is changing with time. In the simulation, comparisons with other high-accuracy methods are also shown. Due to its accuracy and simplicity, the proposed method is practically useful.

Parameter-optimized cubic convolution is used to accurately analyze the pitch center, rate and extent of vibrato tones. We interpolate the time-tracing fundamental frequencies of vibrato tones using parametric cubic convolution, and analytically estimate the positions and values of the extrema, which are used to analyze the characteristics of the vibrato. The optimal values of α, the parameter of the interpolation kernel, are also shown as a function of the normalized vibrato rates.