"Generalized
Raised-Cosine Filters", IEEE Trans. Commun., vol. 47, no. 7, pp.
989-997, June 1999.N. Sheikholeslami Alagha and P. Kabal
"Generalized
Raised-Cosine Filters", IEEE Trans. Commun., vol. 47, no. 7, pp.
989-997, June 1999.
Data transmission over bandlimited channels requires pulse shaping to eliminate or control Intersymbol Interference (ISI). Nyquist filters provide ISI-free transmission. Here we introduce a phase compensation technique to design Nyquist filters. Phase compensation can be applied to the square-root of any zero-phase bandlimited Nyquist filter with normalized excess bandwidth less than or equal to one. The resulting phase compensated square-root filter is also a Nyquist filter. In the case of a raised-cosine spectrum the phase compensator has a simple piecewise linear form. Such a technique is particularly useful to accommodate two different structures for the receiver, one with a filter matched to the transmitting filter and one without a matched filter.
We also use the phase compensation technique to characterize a more general family of Nyquist filters which subsumes raised-cosine spectra. These generalized raised-cosine filters offer more flexibility in filter design. For instance, the rate of asymptotic decay of the filter impulse response may be increased, or the residual ISI introduced by truncation of the impulse response may be minimized. Design example are provided to illustrate these choices.
S. Valaee, B. Champagne, and P. Kabal
"Localization
of Wideband Signals Using Least-Squares and Total Least-Squares Approaches",
IEEE Trans. Signal Processing, vol. 47, no. 5, pp. 1213-1222, May 1999.
In this paper, we introduce a new focusing technique for localization of wideband signals. Relaxing the unitary assumption for the focusing matrices, we formulate the least-square (LS) and the total least-square (TLS) coherent signal-subspace methods. The TLS is an alternative to the conventional LS and uses the fact that the errors can exist both in the focusing location matrix as well as in the estimated location matrix at a given frequency bin. To prevent the focusing loss, we use a class of focusing matrices which are constant under multiplication by their Hermitian transpose. The class of unitary matrices comports with this property. We then develop a new focusing technique based on a modification to the TLS (MTLS). It is shown that the computational complexity of the new technique is significantly lower than that for the rotational signal-subspace method (RSS). The focusing gain of the new technique is also larger than the focusing gain of the RSS algorithm. The simulation study shows that, compared to the RSS, the new algorithm has a smaller resolution signal-to-noise ratio (SNR).
"Natural-Quality
Background Noise Coding Using Residual Substitution", Proc. European
Conf. Speech Communication and Technology (Budapest), pp. 2359-2362,
Sept. 1999.
Existing approaches to background noise coding at very low bit rates (i.e., below 1 kbps) fail to reproduce the noise with natural quality, resulting in a degradation of the overall perceived quality. In this paper, we propose a novel scheme for natural-quality reduced-rate coding of background acoustic noise in voice communication systems. A better representation of the excitation signal in a noise-synthesis model is achieved by classifying the type of acoustic environment noise. Class-dependent residual substitution is used at the receive side to synthesize background noise that sounds similar to the background noise at the transmit side. The improvement in the quality of synthesized noise during the speech gaps helps in preserving noise continuity between talk spurts and speech pauses, and enhances the perceived quality of a conversation.
"Delay
Estimation for Transform Domain Acoustical Echo Cancellation", Proc.
European Conf. Speech Commun., Technol. (Budapest), pp.
2539-2542, Sept. 1999.
Acoustic echo cancellation can be used to remove talker feedback in hands-free systems. Fast convergence and good tracking capabilities cannot be achieved by classical transform domain adaptive filtering algorithms when the reference signal has a variable rank autocorrelation matrix. During the low rank phases of the speech signal, some of the transform-domain tap coefficients become irrelevant to the adaptation process and stop adapting. When the autocorrelation matrix gains full rank, there will be no longer any "frozen" weights. In this paper, we focus on the DCT-LMS algorithm and present a new method using a DCT based delay estimate from other coefficients to move the frozen weights closer to the optimal point and, consequently, reduce the overall re-convergence time.
"An
Improved Background Noise Coding Mode for Variable Rate Speech Coders",
Proc. IEEE Workshop Speech Coding (Porvoo, Finland), pp. 135-137, June 1999.
In this paper, we present a novel background noise coding scheme for variable rate speech coders. Existing approaches to noise coding at very low bit rates (i.e. below 1 kbps) fail to faithfully reproduce background noise resulting in a degradation of the overall perceptual quality. In our approach, classification of the noise type is used to select the type of excitation to be used at the receiver. To illustrate the benefits of our scheme, we have modified the noise coding mode of the CDMA enhanced variable rate codec (EVRC) to include the proposed class-dependent noise excitation model. Evaluation tests have shown that we have improved the overall quality with the proposed noise coding scheme without an increase in bit rate.
H. Najafzadeh-Azghandi and P. Kabal
"Improving
Perceptual Coding of Narrowband Audio Signals at Low Rates", Proc. IEEE
Int. Conf. Acoustics, Speech, Signal Processing (Phoenix, AZ), pp. 913-916,
March 1999.
This paper discusses perceptual coding of narrowband audio signals at low rates. In particular, it proposes a new error measure which shapes the noise inside the critical bands, a window switching criterion based on the temporal masking effect of the hearing system, a more accurate model of the simultaneous masking effect of the hearing system, perceptually-based bit allocation algorithms based on two different approaches towards quantization noise shaping and a predictive vector quantization scheme to code the scale factors. The resulting coding scheme outperforms existing low rate speech coders for non-speech signals.
K. El-Maleh, A. Samouelian, and P. Kabal
"Frame-Level
Noise Classification in Mobile Environments", Proc. IEEE Int. Conf.
Acoustics, Speech, Signal Processing (Phoenix, AZ), pp. 237-240, March 1999.
Background environmental noises degrade the performance of speech-processing systems (e.g. speech coding, speech recognition). By modifying the processing according to the type of background noise, the performance can be enhanced. This requires noise classification. In this paper, four pattern-recognition frameworks have been used to design noise classification algorithms. Classification is done on a frame-by-frame basis (e.g. once every 20 ms). Five commonly encountered noises in mobile telephony (i.e. car, street, babble, factory, and bus) have been considered in our study. Our experimental results show that the Line Spectral Frequencies (LSF's) are robust features in distinguishing the different classes of noises.
Paper titles.