Raymond N. J. Veldhuis

System for subband coding of a digital audio signal

A system for subband coding of a digital audio signal x(k) includes in the coder (1) a filter bank (3) for split ting the audio signal band, with sampling rate reduction, into subbands (p = 1, . . . P) of approximately critical bandwidth and in the decoder (2) a filter bank (5) for merging these subbands, with sampling rate increase. For each subband (p) the coder (1) comprises a detector (7(p)) for determining a parameter G(p;m) representa tive of the signal level in a block (p;m) of Msamples of the subband signal x(k) as well as a quantizer (80p)) for adaptively block quantizing this subband signal in re sponse to parameter G(p;m), and the decoder (2) com prises a dequantizer (9(p)) for adaptively block dequan tizing the quantized subband signal s(k) in response to parameter G(p;m). The quantizing characteristics are related to the noise-masking curve of the human audi tory system, owing to which a high-quality of the rep lica 3(k) of audio signal x(k) is attained with an average number of approximately 2.5 bits per sample for repre senting the oitput signals of the coder (1). The occa sional audibility of quantizing noise in this replica 3(k) is reduced effectively in that the coder (1) and decoder (2) contain identical bit allocation means (23, 24) respon sive to a set of parameters G(p;m) for the higher group of subbands (pin Sps P) within an allocation window (FIG. 5) for allocating a number of B(p;m) bits per sample from a fixed predetermined number of B bits for this allocation window to the quantizer (8(p)) and the dequantizer (9(p)) for the block (p;m) of subband signal x(k) and sp(k), respectively.

Subband coding of stereophonic digital audio signals

The exploitation of left-right correlation in a subband code for stereophonic audio signals is investigated. A transform of left and right signals into decorrelated intensity and error signals is presented. Although this can be seen as the optimal exploitation of redundancy, it yields only marginal gain in bit rate. If the reduced phase-sensitivity of the human observer can be exploited by encoding only the intensity signal, a substantial gain can be obtained. Preliminary results of a stereo codec are promising: at 192 kb/s good coding results have been obtained.<<ETX>>

System for subband coding of a digital audio signal and coder and decoder constituting the same

A system for subband coding of a digital audio signal x(k) includes in the coder (1) a filter bank (3) for splitting the audio signal band, with sampling rate reduction, into subbands (p=1, . . . P) of approximately critical bandwidth and in the decoder (2) a filter bank (5) for merging these subbands, with sampling rate increase. For each subband (p) the coder (1) comprises a detector (7(p)) for determining a parameter G(p;m) representative of the signal level in a block (p;m) of M samples of the subband signal xp (k) as well as a quantizer (8(p)) for adaptively block quantizing this subband signal in response to parameter G(p;m), and the decoder (2) comprises a dequantizer (9(p)) for adaptively block dequantizing the quantized subband signal sp (k) in response to parameter G(p;m). The quantizing characteristics are related to the noise-masking curve of the human auditory system, owing to which a high-quality of the replica x(k) of audio signal x(k) is attained with an average number of approximately 2.5 bits per sample for representing the output signals of the coder (1). The occasional audibility of quantizing noise in this replica x(k) is reduced effectively in that the coder (1) and decoder (2) contain identical bit allocation means (23, 24) responsive to a set of parameters G(p;m) for the higher group of subbands (pim ≦p≦P) within an allocation window (FIG. 5) for allocating a number of B(p;m) bits per sample from a fixed predetermined number of B bits for this allocation window to the quantizer (8(p)) and the dequantizer (9(p)) for the block (p;m) of subband signal xp (k) and sp (k), respectively.

A computationally efficient alternative for the Liljencrants–Fant model and its perceptual evaluation

An alternative for the Liljencrants-Fant (LF) glottal-pulse model is presented. This alternative is derived from the Rosenberg model. Therefore, it is called the Rosenberg++ model. In the derivation a general framework is used for glottal-pulse models. The Rosenberg++ model is described by the same set of T or R parameters as the LF model but it has the advantage over the LF model that it is computationally more efficient. It is compared with the LF model in a psychoacoustic experiment, from which it is concluded that in a practical situation it is capable of producing synthetic speech which is perceptually equivalent to speech generated with the LF model.

Bit rates in audio source coding

The goal is to introduce and solve the audio coding optimization problem. Psychoacoustic results such as masking and excitation pattern models are combined with results from rate distortion theory to formulate the audio coding optimization problem. The solution of the audio optimization problem is a masked error spectrum, prescribing how quantization noise must be distributed over the audio spectrum to obtain a minimal bit rate and an inaudible coding errors. This result cannot only be used to estimate performance bounds, but can also be directly applied in audio coding systems. Subband coding applications to magnetic recording and transmission are discussed in some detail. Performance bounds for this type of subband coding system are derived. >

Hand-Geometry Recognition Based on Contour Parameters

This paper demonstrates the feasibility of a new method of hand-geometry recognition based on parameters derived from the contour of the hand. The contour is completely determined by the black-and-white image of the hand and can be derived from it by means of simple image-processing techniques. It can be modelled by parameters, or features, that can capture more details of the shape of the hand than what is possible with the standard geometrical features used in hand-geometry recognition. The set of features considered in this paper consists of the spatial coordinates of certain landmarks on the contour. The feature set and the recognition method used are discussed in detail. The usefulness of the proposed feature set is evaluated experimentally in a verification context. The verification performance obtained with contour-based features is compared with the verification performance of other methods described in the literature.

Subband coding of digital audio signals without loss of quality

A subband coding system for high quality digital audio signals is described. To achieve low bit rates at a high quality level, it exploits the simultaneous masking effect of the human ear. It is shown how this effect can be used in an adaptive bit-allocation scheme. The proposed approach has been applied in two coding systems, a complex system in which signal is split into 26 subbands, each approximately one third of an octave wide, and a simpler 20-band system. Both systems have been designed for coding stereophonic 16-bit compact disk signals with a sampling frequency of 44.1 kHz. With the 26-band system high-quality results can be obtained at bit rates of 220 kb/s. With the 20-band system, similar results can be obtained at bit rates of 360 kb/s.<<ETX>>

A Practical Subspace Approach To Landmarking

A probabilistic, maximum aposteriori approach to finding landmarks in a face image is proposed, which provides a theoretical framework for template based landmarkers. One such landmarker, based on a likelihood ratio detector, is discussed in detail. Special attention is paid to training and implementation issues, in order to minimize storage and processing requirements. In particular a fast approximate singular value decomposition method is proposed to speed up the training process and implementation of the landmarker in the Fourier domain is presented that will speed up the search process. A subspace method for outlier correction and an iterative implementation of the landmarker are both shown to improve its accuracy. The impact of carefully tuning the many parameters of the method is illustrated. The method is extensively tested and compared with alternatives.

Interpolating autoregressive processes: a bound on the restoration error

An upper bound is obtained for the restoration error variance of a sample restoration method for autoregressive processes that was presented by A.J.E.M. Janssen et al. (ibid., vol.ASSP-34, p.317-30, Apr. 1986). The upper bound derived is lower if the autoregressive process has poles close to the unit circle of the complex plane. This situation corresponds to a peaky signal spectrum. The bound is valid for the case in which one sample is unknown in a realization of an autoregressive process of arbitrary finite order. >