This paper studies the performance of the aposteriori recursive least squares lattice filter in the presence of a nonstationary chirp signal. The forward and backward partial correlation (PARCOR) coefficients for a Wiener-Hopf optimal filter are shown to be complex conjugates for the general case of a nonstationary input with constant power. Such an optimal filter is compared to a minimum mean square error based least squares lattice adaptive filter. Expressions are found for the behavior of the first stage of the adaptive filter based on the least squares algorithm. For the general n th stage, the PARCOR coefficients of the previous stages are assumed to have attained Wiener-Hopf optimal steady state. The PARCOR coefficients of such a least squares adaptive filter are compared with the optimal coefficients for such a nonstationary input. The optimal lattice filter is seen to track a chirp input without any error, and the tracking lag in such an adaptive filter is due to the least sq...

Autoregressive (AR) modeling by linear prediction (LP) provides the basis of a wide variety of signal processing and communication systems including parametric spectral estimation and system identification. Perhaps the greatest success of linear prediction techniques is to be found in speech analysis and audio coding. In this paper, we first reviewed the general frameworks of predictive signal modeling and investigated various prediction filter structures including the modified linear predictor. We then empirically compared the compression performamce of these prediction filters by applying to the lossless audio compression system. We also applied different filter orders and block lengths for each filter to explore their influence on the compression ratio....

Real-time bidirectional audio applications, like microphones and monitor speakers in live performances, typically require communication systems with minimum latency. When digital transmission with limited bit rate is desired, this poses tight constraints on the algorithmic delay of the audio coding scheme. We present a delay-free approach employing adaptive differential pulse code modulation (ADPCM) and adaptive spectral shaping of the coding noise. To achieve zero-delay operation, both prediction and quantization logic of the ADPCM structure are realized in a backwardadaptive fashion. Noise shaping is accomplished via two feedback loops around the quantizer for efficient exploitation of the auditory selectivity and masking phenomena, respectively. Due to automatic optimization of the involved parameters, the performance of the proposed system is on par with that of prior low-delay approaches. 1.