Channel Encoder Research Articles

Arithmetic coding for data compression

The state of the art in data compression is arithmetic coding, not the better-known Huffman method. Arithmetic coding gives greater compression, is faster for adaptive models, and clearly separates the model from the channel encoding.

Space shuttle communications and tracking system

This paper is intended to provide a comprehensive description of the Space Shuttle communications and tracking system design, development, and system operations. Areas of special innovative communications techniques implemented by the Space Shuttle such as source encoding, channel encoding, spread spectrum, and digital modulation are emphasized. In addition, the role of the Electronics Systems Test Laboratory in the development of the system is discussed.

An Approach to the Implementation of a Discrete Cosine Transform

An approach to the implementation of a discrete cosine transform (DCT) for application to coding speech is described. The approach is oriented toward single speech channel encoding. In addition, a detailed computer simulation of an adaptive transform coder is described. The purpose of the computer simulation is to determine the internal precision at various points in the implementation required to avoid subjective degradation. Specific recommmendations are made on the required internal precision in the implementation of the discrete cosine transform. A breadboard implementation of the DCT using SSI and MSI TTL logic based on the results of the computer simulation is reported.

Graph decomposition: A new key to coding theorems

A new and simple method is proposed for finding good encoders both for channels and for sources with side information. This method relies on the continuous version of a graph decomposition result of Lovász. The presently known best exponential error bounds for both problems follow in a unified manner with an improvement on the source coding bound. The previous bounds for universal codes of the authors and Marton are also improved.

Transparent Wideband Data Transmission in Digital Communication Networks

This paper presents a systematic study of the possibilities of time-transparent and of code-transparent binary data transmission in digital communication networks where PCM or differential PCM (DPCM) systems are employed for the encoding of speech or video signals. To describe the effects that occur if data signals instead of speech or video signals are sampled and transmitted by these two encoding methods, time-transparency and code-transparency are defined and are used as figures of merit. The occurrence of time-quantizing errors and amplitude errors at the decoder output is statistically analyzed. The limitations to be considered in the various applications are described, and time- and code-transparency are evaluated quantitatively. The predictions of the analysis are then compared with the results obtained through simulations. The comparison shows that for data rates below 10 percent of the bit rate (bits per second) in the digital channel, conventional PCM and DPCM encoding are acceptable for binary data transmission and have the advantage of easy signal integration. For higher data rates special data encoding is necessary. The encoding method analyzed here is dual-mode.

Virtual holography: a method of source and channel encoding and decoding of information

A method called virtual holography is developed both theoretically and experimentally. The method can be summarized briefly as registering the hologram image generated by an optical system. The real hologram, which is the hologram physically generated and contains transformed information, is designed according to the constraints of technology and, therefore, represents channel encoding. The virtual hologram, which is the unregistered image of the real hologram and contains the desired information, is designed according to the constraints of the reconstruction and object waves and, therefore, represents source encoding.

A joint source and channel encoding scheme †

A method of joint source and channel encoding using block codes is presented. The method is based on n variation of syndrome encoding. Making use of negacyclic codes, the technique results in both data compression and on error-detection capability for appropriate source probability distributions. Both situations, with and without distortion, are considered. An orthogonal extension of the above scheme is also introduced in the concluding remarks.

Convolutional source encoding

In certain communications problems, such as remote telemetry, it is important that any operations performed at the transmitter be of a simple nature, while operations performed at the receiver can frequently be orders of magnitude more complex. Channel coding is well matched to such situations while conventional source coding is not. To overcome this difficulty of usual source coding, we propose using a convolutional encoder for joint source and channel encoding. When the channel is noiseless this scheme reduces to a convolutional source code that is simpler to encode than any other optimal noiseless source code known to date. In either case, decoding can be a minor variation on sequential decoding.

Relationship between source coding, channel coding, and equalization in data transmission

For data communication through a time-dispersive channel, the optimum channel decoder is shown to be a network which is matched to the channel encoder. In this context, channel decoding precedes equalization in the receiver hierarchy.

Alternatives in digital communications

The following major interrelated aspects in digital communications are described: source encoding, channel encoding and transmission, and receiver methods. Methods for digitizing speech and video are summarized. Digitized voice can be transmitted at 10 kb/s, while phone video requires 1.544 Mb/s. The major aspects of both forward-error control (FEC) and automatic request for repeats (ARQ) are considered. Power-limited forward channels are distinguished from bandwidth-limited ones. Signal distance concepts and maximum-likelihood detection principles provide the framework for describing both block-coded and convolutional-coded signaling. Convolutional coding with Viterbi decoding appears to achieve the most cost-effective forward-error control. The improved FEC techniques benefit the transmission of real-time digitized voice and video and any communication over severely power-llmited links. ARQ methods continue to be effective and practical for most cases where the delay associated with retransmission can be tolerated.

Subjective Effects of Channel Transmission Errors on PCM and DPCM Voice Communication Systems

Isopreference tests were used to study the subjective effects of channel transmission errors on computer-simulated PCM and previous-sample feedback DPCM voice systems. Numerically scaled isopreference contours were obtained and plotted on planes having as axes the number of quantization bits <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N</tex> and the bit error probability <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">p</tex> of the binary symmetric transmission channel. For any <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N</tex> and <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">p</tex> considered, the quality of DPCM speech was found to exceed that of PCM speech. With <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">N</tex> fixed and <tex xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">p</tex> decreasing, the improvement of DPCM relative to PCM increased to the point where quantization noise limited the performance of both systems. Two nonredundant codes, natural binary and folded binary, were used to encode the quantizer output. The two codes yielded comparable speech quality. The maximum subjective quality obtainable for white Gaussian and Rayleigh fading channels was determined as a function of channel capacity. Use of optimum channel encoding was shown to reduce considerably the channel capacity required to obtain a given speech quality. The subjective ratings were compared with measured system signal-to-noise ratios. Finally, the effect of the overload level of the quantizer on the system signal-to-noise ratio was determined.

Simulación e implementación en FPGA de un esquema de codificación del canal sujeto al estandar de Wimax

El presente trabajo describe el diseño, simulación e implementación en un FPGA de un Codificador de Canal para su uso en un sistema Wimax, enfocándose en el estándar IEEE 802.16-2004 el cual representa la implementación fija y forma parte de la investigación en el campo de redes de acceso fijo inalámbrico de banda ancha sin línea de vista. El trabajo presenta las principales características usadas en Wimax para la transmisión y recepción además del funcionamiento de cada uno de los bloques usados para la corrección de errores.Este proyecto presenta una implementación en FPGA que utiliza diseño basado en modelo, y usa el software System Generator junto a Matlab y Simulink, para obtener los datos que nos permitan comprobar el funcionamiento del diseño propuesto de acuerdo a las especificaciones del estándar, y además analizar la capacidad de corrección de errores mediante el uso de curvas BER vs SNR y de las constelaciones a la salida del canal para justificar el uso del sistema diseñado.