CORRELATION PROPERTIES OF QUANTIZATION NOISE

This paper examines the correlation properties of quantization noise. The quantization noise energy is subtractive if the quantizer output levels are optimized for the probability density of the input signal (pdf optimized). This paper gives a new result that shows that a quantizer (uniform or not) which has quantizer break points midway between output levels (a minimum distance quantizer) and is scaled to minimize the mean-square error, also has this property. Examples are shown that show the correlation properties which determine whether the quantization noise energy is subtractive or additive. This paper also considers a postfilter configuration that compensates for the quantization noise. The postfilter frequency domain gains take the correlation properties of the quantization noise into account. An experiment on reducing the effect of quantization noise in speech gives an indication that taking account of the correlation is useful. Index Terms — Quantizers, quantizer noise correlation 1

SCALABLE AUDIO CODING USING WATERMARKING

A scalable audio coding method is proposed using a technique, Quantization Index Modulation, borrowed from watermarking. Some of the information of each layer output is embedded (watermarked) in the previous layer. This approach leads to a saving in bitrate while keeping the distortion almost unchanged. This makes the scalable coding system more efficient in terms of Rate-Distortion. The results show that the proposed method outperforms the scalable audio coding based on reconstruction error quantization which is used in practical systems such as MPEG-4 AAC

Bandwidth Efficient Transmultiplexers, Part 2: Subband Complements and Performance Aspects

Abstract-This paper examines the performance issues relating to the quadrature amplitude modulation (QAM) and vestigial sideband (VSB) transmultiplexers synthesized in [l]. First, an analysis of the limitations of the configured systems regarding intersymbol interference and crosstalk suppression arising from the use of practical filters is made. Based on these observations, a new design technique for an FIR low-pass prototype that takes the practical degradations into account is formulated. The procedure involves the unconstrained optimization of an error function. A performance evaluation reveals that for four of the five systems, the new method is superior to a minimax approach in that lower intersymbol interference and crosstalk distortions are achieved with a smaller number of filter taps. For the other transmultiplexer, the advantage of the optimized design over the minimax design is in the added flexibility of taking crosstalk into account thereby diminishing the crosstalk distortion. The five transmultiplexers can be converted into new subband systems. We show how the optimized design approach formulated for the transmultiplexers carries over to the new subband systems. I

Comparison of voice activity detection algorithms for wireless personal communications systems

V oice activity detection (VAD) algorithms have become an in tegral part of many of the recently standardized wireless cellular and P ersonal Communications Systems (PCS). In this paper, we present acomparative study of the performance of three recently proposed VAD algorithms under various acoustical background noise conditions. We also propose new ideas to enhance the performance of a VAD algorithm in wireless PCS speech applications. 1

in a low bit-rate CELP speech coder *

The pitch filter in a low bit-rate CELP speech coder has a strong impact on the quality of the reconstructed speech. In this paper we propose a pseudo-multi-tap pitch filter with fewer degrees of freedom than the number of prediction coefficients, but which gives a higher pitch prediction gain and a more appropriate frequency response than a conventional one-tap pitch filter. First, we present an analysis model for the pseudo-multi-tap pitch prediction filter. Then, we introduce a pseudo-multi-tap pitch prediction filter with a fractional pitch lag. The prediction gain of the pseudo-multi-tap pitch filter is compared to that of conventional one-tap and three-tap pitch filters with integer and non-integer pitch lags. A switching configuration is also studied. This filter switches modes depending on the prediction gain. The stability of a pseudo-multi-tap pitch synthesis filter in a CELP coder is considered. We propose a stabilization method with a relaxed stability test. This relaxed test gives better results than a strict stability test. Finally, we have incorporated the pseudo-multi-tap pitch filter into a 4.8 kbit/s CELP speech coder. Both the objective SNR and subjective quality are better than for a conventional one-tap pitch filter. Zusammenfassung Das Sprachgrundfrequenzfilter in einem CELP-Sprachcoder mit geringer Bitrate iibt einen starken Einflul3 auf die rekonstruierte Sprache aus. In diesem Artikel schlagen wir ein pseudo-multi-tap (pseudo Polykoeffizienten

QUANTIZATION NOISE ESTIMATION FOR LOG-PCM

ITU-T G.711.1 is a multirate wideband extension for the wellknown ITU-T G.711 pulse code modulation of voice frequencies. The extended system is fully interoperable with the legacy narrowband one. In the case where the legacy G.711 is used to code a speech signal and G.711.1 is used to decode it, quantization noise may be audible. For this situation, the standard proposes an optional postfilter. The application of postfiltering requires an estimation of the quatization noise. In this paper we review the process of estimating this coding noise and we propose a better noise estimator. Index Terms — Postfilter, quantization noise 1

Speech enhancement using a statistically derived filter mapping

We view the speech enhancement task in two aspects: reduction of the perceptual noise level in degraded speech and reconstruction of the degraded information, which may result in improvement of speech intelligibility. We are also very interested in noiseindependent speech enhancement where test noise environments could differ in intensity from those of algorithm development. To this end, we have developed in this paper an algorithm called Noise-Independent Statistical Spectral Mapping (NISSM) to estimate a speech enhancement Wiener filter. NISSM consists of a noise-resistant transformation, which converts noisy speech to a set of noise-resistant features, and a spectral mapping function, which maps the features to autoregressive spectra of clean speech. We will show that the proposed algorithm effectively reduces noise intensity. When the noise intensity of training differs from that of testing, NISSM outperforms significantly a conventional spectral mapping. The algorithm operates frame-by-frame and is designed for real-time application. The noise interference could be stationary or non-stationary white noise with variable intensity. I