COMM 1.1   Optimal Bit Allocation with Side Information
Paolo Prandoni (LCAV, Ecole Polytechnique Federale de Lausanne), Martin Vetterli (EECS Dept. UC Berkeley, USA)
vol. 5, pp. 2411-2414
For a given set of quantizers and a data vector, the optimal bit allocation in a rate/distortion sense is the sequence of quantizers which minimizes the overall distortion for a given bit budget. In an operational framework, this sequence is dependent on the data realization rather than on its probabilistic model and the cost of describing the sequence itself becomes therefore part of the bit budget. We present an allocation algorithm based on dynamic programming which determines the optimal bit allocation taking into account the side information of describing the structure of the allocation itself; practical simplifications of the algorithm are also presented with respect to coding of continuous data sources.

COMM 1.2   Performance of Multiple Description Coders on a Real Channel
Amy R Reibman, Hamid Jafarkhani (AT&T Labs - Research), Michael T Orchard (Princeton University), Yao Wang (Polytechnic University)
vol. 5, pp. 2415-2418
In this paper we explore the ability of multiple description (MD) source coders to achieve good performance on channels other than ideal MD channels. We examine both the overall system design and compare the performance of a system with MD source coder to that of a more traditional system using a layered source coder. For the memoryless channels we consider, MD source coding cannot achieve acceptable performance for a Gaussian memoryless source without appropriate channel coding. Also, in memoryless channels, a system with MD source coding outperforms a layered source coding system only in very poor channels. The introduction of memory in the channel degrades the performance of both systems equally. Using interleaving to reduce the impact of memory in the channel has more influence on performance than the choice of source coder.

COMM 1.3   Multiple Description Coding via Scaling Rotation Transform
Wenqing Jiang, Antonio Ortega (University of Southern California)
vol. 5, pp. 2419-2422
In this paper, we propose a two-stage transform design technique for Multiple Description Transform Coding. The first stage is the structure design in which we enforce a Scaling-Rotation factorization of the transform and we further constrain the transform for specific channel conditions using the knowledge of the input correlation matrix and the desired output correlation matrix. In the second stage, magnitude design, we find the optimal transform from all admissible transforms given by the structure design using the numerical algorithm proposed by Goyal et al. \cite{Goyal981}. Such a design enables a structured transform framework which reduces both the design and implementation complexities compared to an exhaustive search through the whole space of nonorthogonal transforms. We give two examples to illustrate the design idea, the Scaling-Hadamard transform for equal rate channels and the Scaling-DST transform for sequential protection channels.

COMM 1.4   Unequal Error Protection Methods for Perceptual Audio Coders
Deepen Sinha, Carl-Erik W Sundberg (Bell Laboratories, Lucent Technologies)
vol. 5, pp. 2423-2426
In most source coded bit streams certain bits can be much more sensitive to transmission errors than others. Unequal error protection (UEP) offers a mechanism for matching error protection capability to sensitivity to transmission errors. A UEP system typically has the same average transmission rate as a corresponding equal error protection (EEP) system but offers an improved perceived signal quality at equal channel signal to noise ratio. In this work we introduce methods of UEP to the perceptual audio coder (PAC). An error sensitivity classifier divides the bits in classes of different sensitivity. Different channel codes are then applied to each class. We show how punctured convolutional codes can be used for UEP of the PAC bitstream. Experimental results for channels with uniform as well as non-uniform noise/interference level indicate that the systems with UEP exhibit graceful degradation and extended range for applications auch as digital audio broadcasting (DAB).

COMM 1.5   EMBEDDED JOINT SOURCE-CHANNEL CODING OF SPEECH USING SYMBOL PUNCTURING OF TRELLIS CODES
Alexis P Bernard, Xueting Liu, Richard Wesel, Abeer Alwan (University California, Los Angeles)
vol. 5, pp. 2427-2430
This paper presents an embedded joint source-channel coding scheme of speech. The source coder is an embedded variable bit rate perceptually based sub-band coder producing bits with different error sensitivities. The channel encoder is a Rate Compatible Punctured Trellis code (RCPT) which permits rate variability and unequal error protection by puncturing symbols. Furthermore, RCPT code design naturally incorporates large constellation sizes, allowing high information rate per symbol. The embedded speech coder and the rate compatible puncturing of symbols provide the embeddibility of the joint coding scheme. The coder is robust to acoustic noise and produces good quality speech for a wide range of channel conditions (AWGN or fading), allowing digital transmission of speech with analog-like graceful degradation.

COMM 1.6   ADAPTIVE RATE-DISTORTION-BASED THRESHOLDING: APPLICATION IN JPEG COMPRESSION OF MIXED IMAGES FOR PRINTING
Marcia G Ramos (Cornell University), Ricardo L De Queiroz (Xerox Corp.)
vol. 5, pp. 2431-2434
In this paper, we propose a new technique for transform coding based on rate-distortion (RD) optimized thresholding (i.e. discarding) of wasteful coefficients. The novelty in this proposed algorithm is that the distortion measure is made adaptive. We apply the method to the compression of mixed documents (containing text, natural images, and graphics) using JPEG for printing. Although human visual system's response to compression artifacts varies depending on the region, JPEG applies the same coding algorithm throughout the mixed document. This paper takes advantage of perceptual classification to improve the performance of the standard JPEG implementation via adaptive thresholding, while being compatible with the baseline standard. A computationally efficient classification algorithm is presented, and the improved performance of the classified JPEG coder is verified. Tests demonstrate the method's efficiency compared to regular JPEG and to JPEG using non-adaptive thresholding. The non-stationary nature of distortion perception is true for most signal classes and the same concept can be used elsewhere.

COMM 1.7   Separable Karhunen Loeve Transforms for the Weighted Universal Transform Coding Algorithm
Hanying Feng, Michelle Effros (California Institute of Technology)
vol. 5, pp. 2435-2438
The weighted universal transform code (WUTC) is a two-stage transform code that replaces JPEG's single, non-optimal transform code with a jointly designed collection of transform codes to achieve good performance across a broader class of possible sources. Unfortunately, the performance gains of WUTC are achieved at the expense of significant increases in computational complexity and larger codes. We here present a faster, more space-efficient WUTC algorithm. The new algorithm uses separable coding instead of direct KLT. While separable coding gives performance comparable to that of WUTC, it uses only 1/8 of the floating-point multiplications and 1/32 of storage of direct KLT. Experimental results included in this work compare the performance of new separable WUTC with both the WUTC and other fast variations of that algorithm.

COMM 1.8   A Theoretical Model for Time Code Modulation
Dimitris Kalogiros, Vassilis Stylianakis (Wire Communications Laboratory, Dept. of Electrical and Computer Enginnering, University of Patras, HELLAS)
vol. 5, pp. 2439-2442
The traditional waveform coding techniques for digital communication systems convert the original analog input signal into a digital bit stream using uniform sampling in the time domain, i.e., PCM, DM, ADPCM. In this paper we propose the Time Code Modulation (TCM) technique as an alternative coding scheme, where information is extracted from the signal, only at the time instants when necessary. This results in a variable sampling rate, where its mean value is significantly less than the Nyquist rate. In addition we suggest a general theoretical model for TCM and we present simulation results for various implementations of TCM coders and decoders. A theoretical estimation of SNR vs. sampling rate performance is also presented.

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Last Update: 5:26 GMT, March 24, 1999 Ingo Höntsch