A synchronization system for a television bandwidth compression scheme /

COO 1469-0194.

The VISTA system for compression of television signal bandwidths.

Vita.

Spectral coding methods for speech compression and speaker identification

This thesis investigates aspects of encoding the speech spectrum at low bit rates, with extensions to the effect of such coding on automatic speaker identification. Vector quantization (VQ) is a technique for jointly quantizing a block of samples at once, in order to reduce the bit rate of a coding system. The major drawback in using VQ is the complexity of the encoder. Recent research has indicated the potential applicability of the VQ method to speech when product code vector quantization (PCVQ) techniques are utilized. The focus of this research is the efficient representation, calculation and utilization of the speech model as stored in the PCVQ codebook. In this thesis, several VQ approaches are evaluated, and the efficacy of two training algorithms is compared experimentally. It is then shown that these productcode vector quantization algorithms may be augmented with lossless compression algorithms, thus yielding an improved overall compression rate. An approach using a statistical model for the vector codebook indices for subsequent lossless compression is introduced. This coupling of lossy compression and lossless compression enables further compression gain. It is demonstrated that this approach is able to reduce the bit rate requirement from the current 24 bits per 20 millisecond frame to below 20, using a standard spectral distortion metric for comparison. Several fast-search VQ methods for use in speech spectrum coding have been evaluated. The usefulness of fast-search algorithms is highly dependent upon the source characteristics and, although previous research has been undertaken for coding of images using VQ codebooks trained with the source samples directly, the product-code structured codebooks for speech spectrum quantization place new constraints on the search methodology. The second major focus of the research is an investigation of the effect of lowrate spectral compression methods on the task of automatic speaker identification. The motivation for this aspect of the research arose from a need to simultaneously preserve the speech quality and intelligibility and to provide for machine-based automatic speaker recognition using the compressed speech. This is important because there are several emerging applications of speaker identification where compressed speech is involved. Examples include mobile communications where the speech has been highly compressed, or where a database of speech material has been assembled and stored in compressed form. Although these two application areas have the same objective - that of maximizing the identification rate - the starting points are quite different. On the one hand, the speech material used for training the identification algorithm may or may not be available in compressed form. On the other hand, the new test material on which identification is to be based may only be available in compressed form. Using the spectral parameters which have been stored in compressed form, two main classes of speaker identification algorithm are examined. Some studies have been conducted in the past on bandwidth-limited speaker identification, but the use of short-term spectral compression deserves separate investigation. Combining the major aspects of the research, some important design guidelines for the construction of an identification model when based on the use of compressed speech are put forward.

Demonstration of image compression in a low-bandwidth wireless camera network

This paper presents an overview of our demonstration of a low-bandwidth, wireless camera network where image compression is undertaken at each node. We briefly introduce the Fleck hardware platform we have developed as well as describe the image compression algorithm which runs on individual nodes. The demo will show real-time image data coming back to base as individual camera nodes are added to the network. Copyright 2007 ACM.

Memory Bandwidth and Power Reduction Using Lossy Reference Frame Compression in Video Encoding

Large external memory bandwidth requirement leads to increased system power dissipation and cost in video coding application. Majority of the external memory traffic in video encoder is due to reference data accesses. We describe a lossy reference frame compression technique that can be used in video coding with minimal impact on quality while significantly reducing power and bandwidth requirement. The low cost transformless compression technique uses lossy reference for motion estimation to reduce memory traffic, and lossless reference for motion compensation (MC) to avoid drift. Thus, it is compatible with all existing video standards. We calculate the quantization error bound and show that by storing quantization error separately, bandwidth overhead due to MC can be reduced significantly. The technique meets key requirements specific to the video encode application. 24-39% reduction in peak bandwidth and 23-31% reduction in total average power consumption are observed for IBBP sequences.

Design and evaluation of an image analysis platform for low-power, low-bandwidth camera networks

We describe the design and evaluation of a platform for networks of cameras in low-bandwidth, low-power sensor networks. In our work to date we have investigated two different DSP hardware/software platforms for undertaking the tasks of compression and object detection and tracking. We compare the relative merits of each of the hardware and software platforms in terms of both performance and energy consumption. Finally we discuss what we believe are the ongoing research questions for image processing in WSNs.

Real-time image streaming over a low-bandwidth wireless camera network

In this paper we describe the recent development of a low-bandwidth wireless camera sensor network. We propose a simple, yet effective, network architecture which allows multiple cameras to be connected to the network and synchronize their communication schedules. Image compression of greater than 90% is performed at each node running on a local DSP coprocessor, resulting in nodes using 1/8th the energy compared to streaming uncompressed images. We briefly introduce the Fleck wireless node and the DSP/camera sensor, and then outline the network architecture and compression algorithm. The system is able to stream color QVGA images over the network to a base station at up to 2 frames per second. © 2007 IEEE.

Protecting encrypted cookies from compression side-channel attacks

Compression is desirable for network applications as it saves bandwidth; however, when data is compressed before being encrypted, the amount of compression leaks information about the amount of redundancy in the plaintext. This side channel has led to successful CRIME and BREACH attacks on web traffic protected by the Transport Layer Security (TLS) protocol. The general guidance in light of these attacks has been to disable compression, preserving confidentiality but sacrificing bandwidth. In this paper, we examine two techniques - heuristic separation of secrets and fixed-dictionary compression|for enabling compression while protecting high-value secrets, such as cookies, from attack. We model the security offered by these techniques and report on the amount of compressibility that they can achieve.

Memory-efficient spatial prediction image compression scheme

In prediction phase, the hierarchical tree structure obtained from the test image is used to predict every central pixel of an image by its four neighboring pixels. The prediction scheme generates the predicted error image, to which the wavelet/sub-band coding algorithm can be applied to obtain efficient compression. In quantization phase, we used a modified SPIHT algorithm to achieve efficiency in memory requirements. The memory constraint plays a vital role in wireless and bandwidth-limited applications. A single reusable list is used instead of three continuously growing linked lists as in case of SPIHT. This method is error resilient. The performance is measured in terms of PSNR and memory requirements. The algorithm shows good compression performance and significant savings in memory. (C) 2006 Elsevier B.V. All rights reserved.

Signal compression through spatial frequency-based motion estimation

The interest in low bit rate video coding has increased considerably. Despite rapid progress in storage density and digital communication system performance, demand for data-transmission bandwidth and storage capacity continue to exceed the capabilities of available technologies. The growth of data-intensive digital audio, video applications and the increased use of bandwidth-limited media such as video conferencing and full motion video have not only sustained the need for efficient ways to encode analog signals, but made signal compression central to digital communication and data-storage technology. In this paper we explore techniques for compression of image sequences in a manner that optimizes the results for the human receiver. We propose a new motion estimator using two novel block match algorithms which are based on human perception. Simulations with image sequences have shown an improved bit rate while maintaining ''image quality'' when compared to conventional motion estimation techniques using the MAD block match criteria.

A high performance scheme for EEG compression using a multichannel model

The amount of data contained in electroencephalogram (EEG) recordings is quite massive and this places constraints on bandwidth and storage. The requirement of online transmission of data needs a scheme that allows higher performance with lower computation. Single channel algorithms, when applied on multichannel EEG data fail to meet this requirement. While there have been many methods proposed for multichannel ECG compression, not much work appears to have been done in the area of multichannel EEG. compression. In this paper, we present an EEG compression algorithm based on a multichannel model, which gives higher performance compared to other algorithms. Simulations have been performed on both normal and pathological EEG data and it is observed that a high compression ratio with very large SNR is obtained in both cases. The reconstructed signals are found to match the original signals very closely, thus confirming that diagnostic information is being preserved during transmission.

An 8-to-1 bit 1-MS/s SAR ADC With VGA and Integrated Data Compression for Neural Recording

Low power consumption per channel and data rate minimization are two key challenges which need to be addressed in future generations of neural recording systems (NRS). Power consumption can be reduced by avoiding unnecessary processing whereas data rate is greatly decreased by sending spike time-stamps along with spike features as opposed to raw digitized data. Dynamic range in NRS can vary with time due to change in electrode-neuron distance or background noise, which demands adaptability. An analog-to-digital converter (ADC) is one of the most important blocks in a NRS. This paper presents an 8-bit SAR ADC in 0.13-mu m CMOS technology along with input and reference buffer. A novel energy efficient digital-to-analog converter switching scheme is proposed, which consumes 37% less energy than the present state-of-the-art. The use of a ping-pong input sampling scheme is emphasized for multichannel input to alleviate the bandwidth requirement of the input buffer. To reduce the data rate, the A/D process is only enabled through the in-built background noise rejection logic to ensure that the noise is not processed. The ADC resolution can be adjusted from 8 to 1 bit in 1-bit step based on the input dynamic range. The ADC consumes 8.8 mu W from 1 V supply at 1 MS/s speed. It achieves effective number of bits of 7.7 bits and FoM of 42.3 fJ/conversion-step.

The wideband high-linearity mixers for cable television tuner application

The wideband high-linearity mixers for a double conversion cable TV tuner is presented. The up-conversion mixer converts the input signal from 100MHz to 1000 MHz to the intermediate frequency (IF) of I GHz above. And the down-conversion mixer converts the frequency back. The degeneration resistors are used to Improve the linearity. The tuner is implemented in a 0.35 mu m SiGe technology. Input power at 1dB compression point can reach +14.23dBm. The lowest noise figure is 17.5dB. The two mixers consume 103mW under a supply voltage of 5 V.

Towards an Example-Based Image Compression Architecture for Video-Conferencing

This paper consists of two major parts. First, we present the outline of a simple approach to very-low bandwidth video-conferencing system relying on an example-based hierarchical image compression scheme. In particular, we discuss the use of example images as a model, the number of required examples, faces as a class of semi-rigid objects, a hierarchical model based on decomposition into different time-scales, and the decomposition of face images into patches of interest. In the second part, we present several algorithms for image processing and animation as well as experimental evaluations. Among the original contributions of this paper is an automatic algorithm for pose estimation and normalization. We also review and compare different algorithms for finding the nearest neighbors in a database for a new input as well as a generalized algorithm for blending patches of interest in order to synthesize new images. Finally, we outline the possible integration of several algorithms to illustrate a simple model-based video-conference system.

Arithmetic bit recycling data compression

La compression des données est la technique informatique qui vise à réduire la taille de l’information pour minimiser l’espace de stockage nécessaire et accélérer la transmission des données dans les réseaux à bande passante limitée. Plusieurs techniques de compression telles que LZ77 et ses variantes souffrent d’un problème que nous appelons la redondance causée par la multiplicité d’encodages. La multiplicité d’encodages (ME) signifie que les données sources peuvent être encodées de différentes manières. Dans son cas le plus simple, ME se produit lorsqu’une technique de compression a la possibilité, au cours du processus d’encodage, de coder un symbole de différentes manières. La technique de compression par recyclage de bits a été introduite par D. Dubé et V. Beaudoin pour minimiser la redondance causée par ME. Des variantes de recyclage de bits ont été appliquées à LZ77 et les résultats expérimentaux obtenus conduisent à une meilleure compression (une réduction d’environ 9% de la taille des fichiers qui ont été compressés par Gzip en exploitant ME). Dubé et Beaudoin ont souligné que leur technique pourrait ne pas minimiser parfaitement la redondance causée par ME, car elle est construite sur la base du codage de Huffman qui n’a pas la capacité de traiter des mots de code (codewords) de longueurs fractionnaires, c’est-à-dire qu’elle permet de générer des mots de code de longueurs intégrales. En outre, le recyclage de bits s’appuie sur le codage de Huffman (HuBR) qui impose des contraintes supplémentaires pour éviter certaines situations qui diminuent sa performance. Contrairement aux codes de Huffman, le codage arithmétique (AC) peut manipuler des mots de code de longueurs fractionnaires. De plus, durant ces dernières décennies, les codes arithmétiques ont attiré plusieurs chercheurs vu qu’ils sont plus puissants et plus souples que les codes de Huffman. Par conséquent, ce travail vise à adapter le recyclage des bits pour les codes arithmétiques afin d’améliorer l’efficacité du codage et sa flexibilité. Nous avons abordé ce problème à travers nos quatre contributions (publiées). Ces contributions sont présentées dans cette thèse et peuvent être résumées comme suit. Premièrement, nous proposons une nouvelle technique utilisée pour adapter le recyclage de bits qui s’appuie sur les codes de Huffman (HuBR) au codage arithmétique. Cette technique est nommée recyclage de bits basé sur les codes arithmétiques (ACBR). Elle décrit le cadriciel et les principes de l’adaptation du HuBR à l’ACBR. Nous présentons aussi l’analyse théorique nécessaire pour estimer la redondance qui peut être réduite à l’aide de HuBR et ACBR pour les applications qui souffrent de ME. Cette analyse démontre que ACBR réalise un recyclage parfait dans tous les cas, tandis que HuBR ne réalise de telles performances que dans des cas très spécifiques. Deuxièmement, le problème de la technique ACBR précitée, c’est qu’elle requiert des calculs à précision arbitraire. Cela nécessite des ressources illimitées (ou infinies). Afin de bénéficier de cette dernière, nous proposons une nouvelle version à précision finie. Ladite technique devienne ainsi efficace et applicable sur les ordinateurs avec les registres classiques de taille fixe et peut être facilement interfacée avec les applications qui souffrent de ME. Troisièmement, nous proposons l’utilisation de HuBR et ACBR comme un moyen pour réduire la redondance afin d’obtenir un code binaire variable à fixe. Nous avons prouvé théoriquement et expérimentalement que les deux techniques permettent d’obtenir une amélioration significative (moins de redondance). À cet égard, ACBR surpasse HuBR et fournit une classe plus étendue des sources binaires qui pouvant bénéficier d’un dictionnaire pluriellement analysable. En outre, nous montrons qu’ACBR est plus souple que HuBR dans la pratique. Quatrièmement, nous utilisons HuBR pour réduire la redondance des codes équilibrés générés par l’algorithme de Knuth. Afin de comparer les performances de HuBR et ACBR, les résultats théoriques correspondants de HuBR et d’ACBR sont présentés. Les résultats montrent que les deux techniques réalisent presque la même réduction de redondance sur les codes équilibrés générés par l’algorithme de Knuth.