Adaptive filters requiring zero multiplications

Journal ArticleThis paper introduces an adaptive filter structure that requires zero multiplications for its implementations. The primary input signals are quantized using DPCM and the DPCM outputs are processed by the adaptive filter. The sign algorithm. We show that if the parameters are chosen properly, hardware implementation of this filter structure requires no multipliers. Under the assumption that the signals are zero mean, wide-sense stationary, and Gaussian random processes, we derive theoretical results for the mean and mean-squared behavior of the filter. A simulation example is presented that shows very good match between theoretical and empirical results

Time and frequency domain algorithms for speech coding

The promise of digital hardware economies (due to recent advances in VLSI technology), has focussed much attention on more complex and sophisticated speech coding algorithms which offer improved quality at relatively low bit rates. This thesis describes the results (obtained from computer simulations) of research into various efficient (time and frequency domain) speech encoders operating at a transmission bit rate of 16 Kbps. In the time domain, Adaptive Differential Pulse Code Modulation (ADPCM) systems employing both forward and backward adaptive prediction were examined. A number of algorithms were proposed and evaluated, including several variants of the Stochastic Approximation Predictor (SAP). A Backward Block Adaptive (BBA) predictor was also developed and found to outperform the conventional stochastic methods, even though its complexity in terms of signal processing requirements is lower. A simplified Adaptive Predictive Coder (APC) employing a single tap pitch predictor considered next provided a slight improvement in performance over ADPCM, but with rather greater complexity. The ultimate test of any speech coding system is the perceptual performance of the received speech. Recent research has indicated that this may be enhanced by suitable control of the noise spectrum according to the theory of auditory masking. Various noise shaping ADPCM configurations were examined, and it was demonstrated that a proposed pre-/post-filtering arrangement which exploits advantageously the predictor-quantizer interaction, leads to the best subjective performance in both forward and backward prediction systems. Adaptive quantization is instrumental to the performance of ADPCM systems. Both the forward adaptive quantizer (AQF) and the backward oneword memory adaptation (AQJ) were examined. In addition, a novel method of decreasing quantization noise in ADPCM-AQJ coders, which involves the application of correction to the decoded speech samples, provided reduced output noise across the spectrum, with considerable high frequency noise suppression. More powerful (and inevitably more complex) frequency domain speech coders such as the Adaptive Transform Coder (ATC) and the Sub-band Coder (SBC) offer good quality speech at 16 Kbps. To reduce complexity and coding delay, whilst retaining the advantage of sub-band coding, a novel transform based split-band coder (TSBC) was developed and found to compare closely in performance with the SBC. To prevent the heavy side information requirement associated with a large number of bands in split-band coding schemes from impairing coding accuracy, without forgoing the efficiency provided by adaptive bit allocation, a method employing AQJs to code the sub-band signals together with vector quantization of the bit allocation patterns was also proposed. Finally, 'pipeline' methods of bit allocation and step size estimation (using the Fast Fourier Transform (FFT) on the input signal) were examined. Such methods, although less accurate, are nevertheless useful in limiting coding delay associated with SRC schemes employing Quadrature Mirror Filters (QMF)

Picture coding in viewdata systems

Viewdata systems in commercial use at present offer the facility for transmitting alphanumeric text and graphic displays via the public switched telephone network. An enhancement to the system would be to transmit true video images instead of graphics. Such a system, under development in Britain at present uses Differential Pulse Code Modulation (DPCM) and a transmission rate of 1200 bits/sec. Error protection is achieved by the use of error protection codes, which increases the channel requirement. In this thesis, error detection and correction of DPCM coded video signals without the use of channel error protection is studied. The scheme operates entirely at the receiver by examining the local statistics of the received data to determine the presence of errors. Error correction is then undertaken by interpolation from adjacent correct or previousiy corrected data. DPCM coding of pictures has the inherent disadvantage of a slow build-up of the displayed picture at the receiver and difficulties with image size manipulation. In order to fit the pictorial information into a viewdata page, its size has to be reduced. Unitary transforms, typically the discrete Fourier transform (DFT), the discrete cosine transform (DCT) and the Hadamard transform (HT) enable lowpass filtering and decimation to be carried out in a single operation in the transform domain. Size reductions of different orders are considered and the merits of the DFT, DCT and HT are investigated. With limited channel capacity, it is desirable to remove the redundancy present in the source picture in order to reduce the bit rate. Orthogonal transformation decorrelates the spatial sample distribution and packs most of the image energy in the low order coefficients. This property is exploited in bit-reduction schemes which are adaptive to the local statistics of the different source pictures used. In some cases, bit rates of less than 1.0 bit/pel are achieved with satisfactory received picture quality. Unlike DPCM systems, transform coding has the advantage of being able to display rapidly a picture of low resolution by initial inverse transformation of the low order coefficients only. Picture resolution is then progressively built up as more coefficients are received and decoded. Different sequences of picture update are investigated to find that which achieves the best subjective quality with the fewest possible coefficients transmitted

Lossless audio coding using adaptive linear prediction

Master'sMASTER OF ENGINEERIN

One- and two-level filter-bank convolvers

In a recent paper, it was shown in detail that in the case of orthonormal and biorthogonal filter banks we can convolve two signals by directly convolving the subband signals and combining the results. In this paper, we further generalize the result. We also derive the statistical coding gain for the generalized subband convolver. As an application, we derive a novel low sensitivity structure for FIR filters from the convolution theorem. We define and derive a deterministic coding gain of the subband convolver over direct convolution for a fixed wordlength implementation. This gain serves as a figure of merit for the low sensitivity structure. Several numerical examples are included to demonstrate the usefulness of these ideas. By using the generalized polyphase representation, we show that the subband convolvers, linear periodically time varying systems, and digital block filtering can be viewed in a unified manner. Furthermore, the scheme called IFIR filtering is shown to be a special case of the convolver

Performance analysis of adaptive filters equipped with the dual sign algorithm

Journal ArticleAdaptive filters equipped with the sign algorithm are attractive in many applications because of their computational simplicity. Unfortunately, their slow speed of convergence is a major limitation. The dual sign algorithm (DSA) is a means by which the convergence speed can be increased without overly degrading the steady-state performance and with a minimal amount of additional computational complexity. This paper presents a convergence analysis for adaptive filters equipped with the dual sign algorithm. Previous analyses of the dual sign algorithm were based on two assumptions: 1) the input sequence to the adaptive filter is white; 2) the behavior for the DSA is such that it switches from an adaptive filter equipped with the sign algorithm with a relatively large convergence constant to another one with a smaller convergence constant a certain amount of the after the filter is initialized. Both these restrictions are removed for Gaussian input signals in our analysis. A simulation example that shows good match between theoretical and empirical results is also presented in this paper

A DWT based perceptual video coding framework: concepts, issues and techniques

The work in this thesis explore the DWT based video coding by the introduction of a novel DWT (Discrete Wavelet Transform) / MC (Motion Compensation) / DPCM (Differential Pulse Code Modulation) video coding framework, which adopts the EBCOT as the coding engine for both the intra- and the inter-frame coder. The adaptive switching mechanism between the frame/field coding modes is investigated for this coding framework. The Low-Band-Shift (LBS) is employed for the MC in the DWT domain. The LBS based MC is proven to provide consistent improvement on the Peak Signal-to-Noise Ratio (PSNR) of the coded video over the simple Wavelet Tree (WT) based MC. The Adaptive Arithmetic Coding (AAC) is adopted to code the motion information. The context set of the Adaptive Binary Arithmetic Coding (ABAC) for the inter-frame data is redesigned based on the statistical analysis. To further improve the perceived picture quality, a Perceptual Distortion Measure (PDM) based on human vision model is used for the EBCOT of the intra-frame coder. A visibility assessment of the quantization error of various subbands in the DWT domain is performed through subjective tests. In summary, all these findings have solved the issues originated from the proposed perceptual video coding framework. They include: a working DWT/MC/DPCM video coding framework with superior coding efficiency on sequences with translational or head-shoulder motion; an adaptive switching mechanism between frame and field coding mode; an effective LBS based MC scheme in the DWT domain; a methodology of the context design for entropy coding of the inter-frame data; a PDM which replaces the MSE inside the EBCOT coding engine for the intra-frame coder, which provides improvement on the perceived quality of intra-frames; a visibility assessment to the quantization errors in the DWT domain