ATDRSS 300 MB/S modem program

The 300 Mbps modem was developed for direct application to the next generation high data rate Time Division Multiple Access (TDMA) communication system. This modem utilizes continuous phase modulation combined with a restricted range Reed-Solomon Codec to achieve a bandwidth efficiency of 3 bits/sec/hz. The constant envelope amplitude signal allows one to operate the power amplifier in its saturation mode without significant spectral regrowth or bit error rate degradation

Design of adaptive analog filters for magnetic front-end read channels

Esta tese estuda o projecto e o comportamento de filtros em tempo contínuo de muito-alta-frequência. A motivação deste trabalho foi a investigação de soluções de filtragem para canais de leitura em sistemas de gravação e reprodução de dados em suporte magnético, com custos e consumo (tamanho total inferior a 1 mm2 e consumo inferior a 1mW/polo), inferiores aos circuitos existentes. Nesse sentido, tal como foi feito neste trabalho, o rápido desenvolvimento das tecnologias de microelectrónica suscitou esforços muito significativos a nível mundial com o objectivo de se investigarem novas técnicas de realização de filtros em circuito integrado monolítico, especialmente em tecnologia CMOS (Complementary Metal Oxide Semiconductor). Apresenta-se um estudo comparativo a diversos níveis hierárquicos do projecto, que conduziu à realização e caracterização de soluções com as características desejadas. Num primeiro nível, este estudo aborda a questão conceptual da gravação e transmissão de sinal bem como a escolha de bons modelos matemáticos para o tratamento da informação e a minimização de erro inerente às aproximações na conformidade aos princípios físicos dos dispositivos caracterizados. O trabalho principal da tese é focado nos níveis hierárquicos da arquitectura do canal de leitura e da realização em circuito integrado do seu bloco principal – o bloco de filtragem. Ao nível da arquitectura do canal de leitura, apresenta-se um estudo alargado sobre as metodologias existentes de adaptação de sinal e recuperação de dados em suporte magnético. Este desígnio aparece no âmbito da proposta de uma solução de baixo custo, baixo consumo, baixa tensão de alimentação e baixa complexidade, alicerçada em tecnologia digital CMOS, para a realização de um sistema DFE (Decision Feedback Equalization) com base na igualização de sinal utilizando filtros integrados analógicos em tempo contínuo. Ao nível do projecto de realização do bloco de filtragem e das técnicas de implementação de filtros e dos seus blocos constituintes em circuito integrado, concluiu-se que a técnica baseada em circuitos de transcondutância e condensadores, também conhecida como filtros gm-C (ou transcondutância-C), é a mais adequada para a realização de filtros adaptativos em muito-alta-frequência. Definiram-se neste nível hierárquico mais baixo, dois subníveis de aprofundamento do estudo no âmbito desta tese, nomeadamente: a pesquisa e análise de estruturas ideais no projecto de filtros recorrendo a representações no espaço de estados; e, o estudo de técnicas de realização em tecnologia digital CMOS de circuitos de transcondutância para a implementação de filtros integrados analógicos em tempo contínuo. Na sequência deste estudo, apresentam-se e comparam-se duas estruturas de filtros no espaço de estados, correspondentes a duas soluções alternativas para a realização de um igualador adaptativo realizado por um filtro contínuo passa-tudo de terceira ordem, para utilização num canal de leitura de dados em suporte magnético. Como parte constituinte destes filtros, apresenta-se uma técnica de realização de circuitos de transcondutância, e de realização de condensadores lineares usando matrizes de transístores MOSFET para processamento de sinal em muito-alta-frequência realizada em circuito integrado usando tecnologia digital CMOS submicrométrica. Apresentam-se métodos de adaptação automática capazes de compensar os erros face aos valores nominais dos componentes, devidos às tolerâncias inerentes ao processo de fabrico, para os quais apresentamos os resultados de simulação e de medição experimental obtidos. Na sequência deste estudo, resultou igualmente a apresentação de um circuito passível de constituir uma solução para o controlo de posicionamento da cabeça de leitura em sistemas de gravação/reprodução de dados em suporte magnético. O bloco proposto é um filtro adaptativo de primeira ordem, com base nos mesmos circuitos de transcondutância e técnicas de igualação propostos e utilizados na implementação do filtro adaptativo de igualação do canal de leitura. Este bloco de filtragem foi projectado e incluído num circuito integrado (Jaguar) de controlo de posicionamento da cabeça de leitura realizado para a empresa ATMEL em Colorado Springs, e incluído num produto comercial em parceria com uma empresa escocesa utilizado em discos rígidos amovíveis.This thesis studies the design and behavior of continuous-time very-high-frequency filters. The motivation of this work was the search for filtering solutions for the readchannel in recording and reproduction of data on magnetic media systems, with costs and consumption (total size less than 1 mm2 and consumption under 1mW/pole), lower than the available circuits. Accordingly, as was done in this work, the rapid development of microelectronics technology raised very significant efforts worldwide in order to investigate new techniques for implementing such filters in monolithic integrated circuit, especially in CMOS technology (Complementary Metal Oxide Semiconductor). We present a comparative study on different hierarchical levels of the project, which led to the realization and characterization of solutions with the desired characteristics. In the first level, this study addresses the conceptual question of recording and transmission of signal and the choice of good mathematical models for the processing of information and minimization of error inherent in the approaches and in accordance with the principles of the characterized physical devices. The main work of this thesis is focused on the hierarchical levels of the architecture of the read channel and the integrated circuit implementation of its main block - the filtering block. At the architecture level of the read channel this work presents a comprehensive study on existing methodologies of adaptation and signal recovery of data on magnetic media. This project appears in the sequence of the proposed solution for a lowcost, low consumption, low voltage, low complexity, using CMOS digital technology for the performance of a DFE (Decision Feedback Equalization) based on the equalization of the signal using integrated analog filters in continuous time. At the project level of implementation of the filtering block and techniques for implementing filters and its building components, it was concluded that the technique based on transconductance circuits and capacitors, also known as gm-C filters is the most appropriate for the implementation of very-high-frequency adaptive filters. We defined in this lower level, two sub-levels of depth study for this thesis, namely: research and analysis of optimal structures for the design of state-space filters, and the study of techniques for the design of transconductance cells in digital CMOS circuits for the implementation of continuous time integrated analog filters. Following this study, we present and compare two filtering structures operating in the space of states, corresponding to two alternatives for achieving a realization of an adaptive equalizer by the use of a continuous-time third order allpass filter, as part of a read-channel for magnetic media devices. As a constituent part of these filters, we present a technique for the realization of transconductance circuits and for the implementation of linear capacitors using arrays of MOSFET transistors for signal processing in very-high-frequency integrated circuits using sub-micrometric CMOS technology. We present methods capable of automatic adjustment and compensation for deviation errors in respect to the nominal values of the components inherent to the tolerances of the fabrication process, for which we present the simulation and experimental measurement results obtained. Also as a result of this study, is the presentation of a circuit that provides a solution for the control of the head positioning on recording/playback systems of data on magnetic media. The proposed block is an adaptive first-order filter, based on the same transconductance circuits and equalization techniques proposed and used in the implementation of the adaptive filter for the equalization of the read channel. This filter was designed and included in an integrated circuit (Jaguar) used to control the positioning of the read-head done for ATMEL company in Colorado Springs, and part of a commercial product used in removable hard drives fabricated in partnership with a Scottish company

Bit-level pipelined digit-serial array processors

A new architecture for high performance digit-serial vector inner product (VIP) which can be pipelined to the bit-level is introduced. The design of the digit-serial vector inner product is based on a new systematic design methodology using radix-2n arithmetic. The proposed architecture allows a high level of bit-level pipelining to increase the throughput rate with minimum initial delay and minimum area. This will give designers greater flexibility in finding the best tradeoff between hardware cost and throughput rate. It is shown that sub-digit pipelined digit-serial structure can achieve a higher throughput rate with much less area consumption than an equivalent bit-parallel structure. A twin-pipe architecture to double the throughput rate of digit-serial multipliers and consequently that of the digit-serial vector inner product is also presented. The effect of the number of pipelining levels and the twin-pipe architecture on the throughput rate and hardware cost are discussed. A two's complement digit-serial architecture which can operate on both negative and positive numbers is also presented

Transmitter precoding and code-sharing techniques using block transmission system [TK1-9971].

Dewasa ini, perkhidmatan komunikasi bergerak telah meledak dengan begitu pantas sekali dalam masyarakat kita. Kesemua sistem komunikasi selular yang ada kini menggunakan teknologi digital. Recently, mobile communications services are penetrating into our society at an explosive growth rate. All of the current cellular communication systems have adopted digital technology

Advanced Modulation and Coding Technology Conference

The objectives, approach, and status of all current LeRC-sponsored industry contracts and university grants are presented. The following topics are covered: (1) the LeRC Space Communications Program, and Advanced Modulation and Coding Projects; (2) the status of four contracts for development of proof-of-concept modems; (3) modulation and coding work done under three university grants, two small business innovation research contracts, and two demonstration model hardware development contracts; and (4) technology needs and opportunities for future missions

A low-complexity equalizer for video broadcasting in cyber-physical social systems through handheld mobile devices

In Digital Video Broadcasting-Handheld (DVB-H) devices for cyber-physical social systems, the Discrete Fractional Fourier Transform-Orthogonal Chirp Division Multiplexing (DFrFT-OCDM) has been suggested to enhance the performance over Orthogonal Frequency Division Multiplexing (OFDM) systems under time and frequency-selective fading channels. In this case, the need for equalizers like the Minimum Mean Square Error (MMSE) and Zero-Forcing (ZF) arises, though it is excessively complex due to the need for a matrix inversion, especially for DVB-H extensive symbol lengths. In this work, a low complexity equalizer, Least-Squares Minimal Residual (LSMR) algorithm, is used to solve the matrix inversion iteratively. The paper proposes the LSMR algorithm for linear and nonlinear equalizers with the simulation results, which indicate that the proposed equalizer has significant performance and reduced complexity over the classical MMSE equalizer and other low complexity equalizers, in time and frequency-selective fading channels. © 2013 IEEE

FPGA-based DOCSIS upstream demodulation

In recent years, the state-of-the-art in field programmable gate array (FPGA) technology has been advancing rapidly. Consequently, the use of FPGAs is being considered in many applications which have traditionally relied upon application-specific integrated circuits (ASICs). FPGA-based designs have a number of advantages over ASIC-based designs, including lower up-front engineering design costs, shorter time-to-market, and the ability to reconfigure devices in the field. However, ASICs have a major advantage in terms of computational resources. As a result, expensive high performance ASIC algorithms must be redesigned to fit the limited resources available in an FPGA. Concurrently, coaxial cable television and internet networks have been undergoing significant upgrades that have largely been driven by a sharp increase in the use of interactive applications. This has intensified demand for the so-called upstream channels, which allow customers to transmit data into the network. The format and protocol of the upstream channels are defined by a set of standards, known as DOCSIS 3.0, which govern the flow of data through the network. Critical to DOCSIS 3.0 compliance is the upstream demodulator, which is responsible for the physical layer reception from all customers. Although upstream demodulators have typically been implemented as ASICs, the design of an FPGA-based upstream demodulator is an intriguing possibility, as FPGA-based demodulators could potentially be upgraded in the field to support future DOCSIS standards. Furthermore, the lower non-recurring engineering costs associated with FPGA-based designs could provide an opportunity for smaller companies to compete in this market. The upstream demodulator must contain complicated synchronization circuitry to detect, measure, and correct for channel distortions. Unfortunately, many of the synchronization algorithms described in the open literature are not suitable for either upstream cable channels or FPGA implementation. In this thesis, computationally inexpensive and robust synchronization algorithms are explored. In particular, algorithms for frequency recovery and equalization are developed. The many data-aided feedforward frequency offset estimators analyzed in the literature have not considered intersymbol interference (ISI) caused by micro-reflections in the channel. It is shown in this thesis that many prominent frequency offset estimation algorithms become biased in the presence of ISI. A novel high-performance frequency offset estimator which is suitable for implementation in an FPGA is derived from first principles. Additionally, a rule is developed for predicting whether a frequency offset estimator will become biased in the presence of ISI. This rule is used to establish a channel excitation sequence which ensures the proposed frequency offset estimator is unbiased. Adaptive equalizers that compensate for the ISI take a relatively long time to converge, necessitating a lengthy training sequence. The convergence time is reduced using a two step technique to seed the equalizer. First, the ISI equivalent model of the channel is estimated in response to a specific short excitation sequence. Then, the estimated channel response is inverted with a novel algorithm to initialize the equalizer. It is shown that the proposed technique, while inexpensive to implement in an FPGA, can decrease the length of the required equalizer training sequence by up to 70 symbols. It is shown that a preamble segment consisting of repeated 11-symbol Barker sequences which is well-suited to timing recovery can also be used effectively for frequency recovery and channel estimation. By performing these three functions sequentially using a single set of preamble symbols, the overall length of the preamble may be further reduced

Efficient channel equalization algorithms for multicarrier communication systems

Blind adaptive algorithm that updates time-domain equalizer (TEQ) coefficients by Adjacent Lag Auto-correlation Minimization (ALAM) is proposed to shorten the channel for multicarrier modulation (MCM) systems. ALAM is an addition to the family of several existing correlation based algorithms that can achieve similar or better performance to existing algorithms with lower complexity. This is achieved by designing a cost function without the sum-square and utilizing symmetrical-TEQ property to reduce the complexity of adaptation of TEQ to half of the existing one. Furthermore, to avoid the limitations of lower unstable bit rate and high complexity, an adaptive TEQ using equal-taps constraints (ETC) is introduced to maximize the bit rate with the lowest complexity. An IP core is developed for the low-complexity ALAM (LALAM) algorithm to be implemented on an FPGA. This implementation is extended to include the implementation of the moving average (MA) estimate for the ALAM algorithm referred as ALAM-MA. Unit-tap constraint (UTC) is used instead of unit-norm constraint (UNC) while updating the adaptive algorithm to avoid all zero solution for the TEQ taps. The IP core is implemented on Xilinx Vertix II Pro XC2VP7-FF672-5 for ADSL receivers and the gate level simulation guaranteed successful operation at a maximum frequency of 27 MHz and 38 MHz for ALAM-MA and LALAM algorithm, respectively. FEQ equalizer is used, after channel shortening using TEQ, to recover distorted QAM signals due to channel effects. A new analytical learning based framework is proposed to jointly solve equalization and symbol detection problems in orthogonal frequency division multiplexing (OFDM) systems with QAM signals. The framework utilizes extreme learning machine (ELM) to achieve fast training, high performance, and low error rates. The proposed framework performs in real-domain by transforming a complex signal into a single 2–tuple real-valued vector. Such transformation offers equalization in real domain with minimum computational load and high accuracy. Simulation results show that the proposed framework outperforms other learning based equalizers in terms of symbol error rates and training speeds

Modeling and Design of Architectures for High-Speed ADC-Based Serial Links

There is an ongoing dramatic rise in the volume of internet traffic. Standards such as 56Gb/s OIF very short reach (VSR), medium reach (MR) and long reach (LR) standards for chip to chip communication over channels with up to 10dB, 20dB and 30dB insertion loss at the PAM 4 Nyquist frequency, respectively, are being adopted. These standards call for the spectrally efficient PAM-4 signaling over NRZ signaling. PAM-4 signaling offers challenges such as a reduced SNR at the receiver, susceptibility to nonlinearities and increased sensitivity to residual ISI. Equalization provided by traditional mixed signal architectures can be insufficient to achieve the target BER requirements for very long reach channels. ADC-based receiver architectures for PAM-4 links take advantage of the more powerful equalization techniques, which lend themselves to easier and robust digital implementations, to extend the amount of insertion loss that the receiver can handle. However, ADC-based receivers can consume more power compared to mixed-signal implementations. Techniques that model the receiver performance to understand the various system trade-offs are necessary. This research presents a fast and accurate hybrid modeling framework to efficiently investigate system trade-offs for an ADC-based receiver. The key contribution being the addition of ADC related non-idealities such as quantization noise in the presence of integral and differential nonlinearities, and time-interleaving mismatch errors such as gain mismatch, bandwidth mismatch, offset mismatch and sampling skew. The research also presents a 52Gb/s ADC-based PAM-4 receiver prototype employing a 32-way time-interleaved, 2-bit/stage, 6-bit SAR ADC and a DSP with a 12-tap FFE and a 2-tap DFE. A new DFE architecture that reduces the complexity of a PAM-4 DFE to that of an NRZ DFE while simultaneously nearly doubling the maximum achievable data rate is presented. The receiver architecture also includes an analog front-end (AFE) consisting of a programmable two stage CTLE. A digital baud-rate CDR’s utilizing a Mueller-Muller phase detector sets the sampling phase. Measurement results show that for 32Gb/s operation a BER < 10⁻⁹ is achieved for a 30dB loss channel while for 52 Gb/s operation achieves a BER < 10⁻⁶ for a 31dB loss channel with a power efficiency of 8.06pj/bit

Design of Energy-Efficient A/D Converters with Partial Embedded Equalization for High-Speed Wireline Receiver Applications

As the data rates of wireline communication links increases, channel impairments such as skin effect, dielectric loss, fiber dispersion, reflections and cross-talk become more pronounced. This warrants more interest in analog-to-digital converter (ADC)-based serial link receivers, as they allow for more complex and flexible back-end digital signal processing (DSP) relative to binary or mixed-signal receivers. Utilizing this back-end DSP allows for complex digital equalization and more bandwidth-efficient modulation schemes, while also displaying reduced process/voltage/temperature (PVT) sensitivity. Furthermore, these architectures offer straightforward design translation and can directly leverage the area and power scaling offered by new CMOS technology nodes. However, the power consumption of the ADC front-end and subsequent digital signal processing is a major issue. Embedding partial equalization inside the front-end ADC can potentially result in lowering the complexity of back-end DSP and/or decreasing the ADC resolution requirement, which results in a more energy-effcient receiver. This dissertation presents efficient implementations for multi-GS/s time-interleaved ADCs with partial embedded equalization. First prototype details a 6b 1.6GS/s ADC with a novel embedded redundant-cycle 1-tap DFE structure in 90nm CMOS. The other two prototypes explain more complex 6b 10GS/s ADCs with efficiently embedded feed-forward equalization (FFE) and decision feedback equalization (DFE) in 65nm CMOS. Leveraging a time-interleaved successive approximation ADC architecture, new structures for embedded DFE and FFE are proposed with low power/area overhead. Measurement results over FR4 channels verify the effectiveness of proposed embedded equalization schemes. The comparison of fabricated prototypes against state-of-the-art general-purpose ADCs at similar speed/resolution range shows comparable performances, while the proposed architectures include embedded equalization as well