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Volume 147
Issue 6
IEE Proceedings - Vision, Image and Signal Processing
Volume 147, Issue 6, December 2000
Volumes & issues:
Volume 147, Issue 6
December 2000
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- Author(s): P. Gray ; M.P. Hollier ; R.E. Massara
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 493 –501
- DOI: 10.1049/ip-vis:20000539
- Type: Article
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p.
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It is increasingly vital that there are effective quality of service (QoS) metrics to describe the performance of telecommunications networks. Speech quality is a major contributor to users' perception of QoS, and the ability to design for and monitor this quality is paramount. The authors describe work towards a non-intrusive speech quality assessment algorithm, capable of making predictions of the speech quality received by a customer, utilising the in-service signal. Modern telecommunications networks contain complex nonlinear elements that cannot be assessed with traditional engineering metrics. A novel use of vocal-tract modelling techniques is described, which enables predictions of the quality of a network degraded speech stream to be made. Details of the algorithm's adaptation to different talker characteristics are presented, together with a summary of the performance of the system. - Author(s): M. Li ; H.G. McAllister ; N.D. Black ; T.A. De Perez
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 502 –507
- DOI: 10.1049/ip-vis:20000631
- Type: Article
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Hearing impairment can often be corrected by medical or surgical treatment, provided the loss is not due to cochlear pathology (sensorineural loss). For sensorineural loss, the only corrective action is to wear a hearing aid. Many perceptual differences between normal and sensorineural hearing-impaired listeners are due to differences in the dB levels at which sound is detected by the ear (auditory threshold) and the associated dynamic range over which it is comfortable to listen to (loudness sensation). To adequately compensate for this, the processing of the auditory signal should be nonlinear and time-varying. Two wavelet-based compression algorithms have been developed: automatic gain control (AGC) with linear amplification and nonlinear compression AGC. The nonlinear AGC is a compression algorithm, which models loudness sensation. The wavelet transform separates the input into seven frequency bands corresponding to the critical bands of the human auditory system. For each frequency band, multiplying the wavelet coefficients by the gain can amplify or compress nonlinearly and smoothly, depending on the signal level, time and spectrum. Results suggest that the nonlinear approach, while maintaining the general spectral structure of the signal, is perceptually superior to linear AGC and compensates better for audibility loss. - Author(s): J.-T. Chien and M.-S. Lin
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 508 –515
- DOI: 10.1049/ip-vis:20000693
- Type: Article
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It has become increasingly important to develop hands-free speech recognition techniques for the human–computer interface in car environments. However, severe car noise degrades the speech recognition performance substantially. To compensate the performance loss, it is necessary to adapt the original speech hidden Markov models (HMMs) to meet changing car environments. A novel frame-synchronous adaptation mechanism for in-car speech recognition is presented. This mechanism is intended to perform unsupervised model adaptation efficiently on a frame-by-frame basis instead of a conventional adaptation algorithm relying on batch adaptation data and supervision information. The proposed adaptation scheme is performed during frame likelihood calculation where an optimal equalisation factor is first computed to equalise the model mean vector and the input frame vector. This equalisation factor then serves as a reference index to retrieve an additional bias vector for model mean adaptation. As a result, a rapid and flexible algorithm is exploited to establish a new robust likelihood measure. In experiments on hands-free in-car speech recognition with the microphone far from the talker, this framework is found to be effective in terms of recognition rate and computational cost under various driving speeds. - Author(s): C. Rusu and C.F.N. Cowan
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 516 –526
- DOI: 10.1049/ip-vis:20000595
- Type: Article
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A family of stochastic gradient algorithms and their behaviour in the data echo cancellation work platform are presented. The cost function adaptation algorithms use an error exponent update strategy based on an absolute error mapping, which is updated at every iteration. The quadratic and nonquadratic cost functions are special cases of the new family. Several possible realisations are introduced using these approaches. The noisy error problem is discussed and the digital recursive filter estimator is proposed. The simulation outcomes confirm the effectiveness of the proposed family of algorithms. - Author(s): Y. Zhao and M.N.S. Swamy
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 527 –533
- DOI: 10.1049/ip-vis:20000594
- Type: Article
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A new technique based on nonlinear optimisation to design nearly orthogonal wavelet filter banks with linear phase is proposed. The main idea is to impose a certain number of zeros at z=−1 for a symmetric filter and make it satisfy the power complementary condition as accurately as possible. From this filter, a semi-orthogonal wavelet filter bank which is nearly orthogonal can be constructed. This semi-orthogonal filter bank can be approximately implemented using a filter bank consisting of only one prototype filter. The frequency selectivity can also be designed at the same time by using a weighted cost function. - Author(s): J.-H. Lee and I.-C. Niu
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 534 –542
- DOI: 10.1049/ip-vis:20000489
- Type: Article
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The nonlinear optimisation problem that results from considering the design of a two-channel nonuniform division filter bank is solved. This is through a frequency sampling and iterative approximation technique to find the tap coefficients and the reflection coefficients for the numerator and the denominator of the IIR analysis filters. An efficient stabilisation procedure ensures that the reflection coefficients lie in (−1,1). Simulation examples are provided for illustration. - Author(s): H.H. Dam ; S. Nordebo ; K.L. Teo ; A. Cantoni
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 543 –548
- DOI: 10.1049/ip-vis:20000598
- Type: Article
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The difference routing digital filter (DRDF) consists of an FIR filter followed by a first-order integrator. This structure with power-of-two coefficients has been studied as a means of achieving low complexity, high sampling rate filters which can be implemented efficiently in hardware. The optimisation of the coefficients has previously been based on a time-domain least-squares error criterion. A new design method is proposed that includes a frequency-domain least-squares criterion with arbitrary frequency weighting and an improved method for handling quantisation of the filter coefficients. Simulation studies show that the new approach yields an improvement of up to 7 dB over existing methods and that oversampling can be used to improve performance. - Author(s): S.C. Dutta Roy
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 549 –552
- DOI: 10.1049/ip-vis:20000733
- Type: Article
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The usual procedure for synthesising the Nth order digital FIR transfer function AN(z) in which the constant term is unity and the coefficient of z−N is −aN(N) by a lattice structure fails if aN(N)=±1. The author is concerned with a solution to this problem. It is known that under this condition. AN(z) must be linear phase to be realisable by a lattice structure. A synthesis procedure is given for such an AN(z). For a general nonlinear phase AN(z) with aN(N)=±1, an alternative synthesis procedure is also given in terms of parallel lattice structures. - Author(s): I.R. Khan ; R. Ohba ; N. Hozumi
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 553 –555
- DOI: 10.1049/ip-vis:20000726
- Type: Article
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Explicit formulas for the tap-coefficients of Taylor-series-based full-band FIR digital differentiators have already been presented. However, those formulas were not derived mathematically from the Taylor series and were based on observations of different sets of results. The authors provide a mathematical proof of the formulas by deriving them mathematically from the Taylor series. - Author(s): S. Fiori and G. Maiolini
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 557 –563
- DOI: 10.1049/ip-vis:20000695
- Type: Article
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A new algorithm based on a weighted least-squares technique is proposed that allows online deconvolution of non-minimum phase systems with neither knowledge of the system's impulse response nor source statistics except for source signal moments up to the fourth order. Using computer simulations and computational complexity evaluation, the authors illustrate and compare the performances and the features of the proposed method with those of other techniques found in the literature. - Author(s): J.M. Zhong ; C.H. Leung ; Y.Y. Tang
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 564 –570
- DOI: 10.1049/ip-vis:20000752
- Type: Article
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An efficient image compression algorithm based on energy clustering and zero-quadtree representation (ECZQR) in the wavelet transform domain is proposed. In embedded coding, zeros within each subband are encoded in the framework of quadtree representation instead of zerotree representation. To use large rectangular blocks to represent zeros, it first uses morphological dilation to extract the arbitrarily shaped clusters of significant coefficients within each subband. The proposed encoding method results in less distortion in the decoded image than the line-by-line encoding method. Experimental results show that the algorithm is among the most efficient wavelet image compression algorithms. - Author(s): J.H. Jeng ; T.K. Truong ; J.R. Sheu
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 571 –574
- DOI: 10.1049/ip-vis:20000756
- Type: Article
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p.
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A new algorithm for fractal image compression is developed to speed up the encoder. This new method converts image blocks into frequency-like domains using the Hadamard transform, in which the computations of the best matched are performed. At each search entry, the best mean square error computations of the eight dihedral symmetries are reformulated into the form of inner products. By a precise derivation, all redundant computations are completely avoided. With this improved technique, the complexity of the encoder is substantially reduced. A simulation shows that, with the same PSNR and compression ratio, the new method requires less computation time than the baseline method. - Author(s): J. Jiang ; B. Guo ; S.Y. Yang
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 575 –580
- DOI: 10.1049/ip-vis:20000767
- Type: Article
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The authors investigate the prediction scheme of JPEG-LS, the latest JPEG standard for lossless/near lossless image compression. They show that it is not sufficient to consider only horizontal and vertical edges in constructing predictive values. As a result, they propose an additional diagonal edge detection scheme to achieve better prediction accuracy and hence provide potential for further improvement. Experiments show that, in terms of mean-square-error values, the proposed scheme outperforms the existing JPEG-LS prediction for all images tested, while the complexity of the overall algorithm is maintained at a similar level. - Author(s): L.D. Scargall and S.S. Dlay
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 581 –586
- DOI: 10.1049/ip-vis:20000614
- Type: Article
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An image sequence coding scheme for low bit-rate video coding is presented. A new methodology is proposed for adaptive vector quantisation (AVQ), where the codebook is updated with new code-vectors. The new code-vectors replace less significant ones in the codebook, based on a novel-scoring criterion that utilises a forgetting factor and codebook half-life. The proposed AVQ method gives rise to an additional performance enhancement of approximately 1 dB over frequency scoring techniques. - Author(s): S.O. Choy ; Y.H. Chan ; W.C. Siu
- Source: IEE Proceedings - Vision, Image and Signal Processing, Volume 147, Issue 6, p. 587 –594
- DOI: 10.1049/ip-vis:20000383
- Type: Article
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Conventional spatially adaptive regularised image restoration schemes weight the amount of regularisation according to the spatial content of an image. The authors first separately decorrelate the signals under analysis into uncorrelated components and then weight the amount of regularisation performed to these components accordingly. The proposed approach works better than conventional schemes, especially in edge regions.
Non-intrusive speech-quality assessment using vocal-tract models
Wavelet-based nonlinear AGC method for hearing aid loudness compensation
Frame-synchronous noise compensation for hands-free speech recognition in car environments
Cost function adaptation: a stochastic gradient algorithm for data echo cancellation
New technique for designing nearly orthogonal wavelet filter banks with linear phase
Design of two-channel IIR filter banks with arbitrary group delay
FIR filter design over discrete coefficients and least square error
Synthesis of FIR lattice structures
Mathematical proof of explicit formulas for tap-coefficients of full-band FIR digital differentiators
Weighted least-squares blind deconvolution of non-minimum phase systems
Image compression based on energy clustering and zero-quadtree representation
Fast fractal image compression using the Hadamard transform
Revisiting the JPEG-LS prediction scheme
New methodology for adaptive vector quantisation
Image restoration by regularisation in uncorrelated transform domain
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