Showing papers on "Digital signal processing published in 2008"

Digital filters for coherent optical receivers.

Abstract: Digital filters underpin the performance of coherent optical receivers which exploit digital signal processing (DSP) to mitigate transmission impairments. We outline the principles of such receivers and review our experimental investigations into compensation of polarization mode dispersion. We then consider the details of the digital filtering employed and present an analytical solution to the design of a chromatic dispersion compensating filter. Using the analytical solution an upper bound on the number of taps required to compensate chromatic dispersion is obtained, with simulation revealing an improved bound of 2.2 taps per 1000ps/nm for 10.7GBaud data. Finally the principles of digital polarization tracking are outlined and through simulation, it is demonstrated that 100krad/s polarization rotations could be tracked using DSP with a clock frequency of less than 500MHz.

Coherent detection in optical fiber systems

Abstract: The drive for higher performance in optical fiber systems has renewed interest in coherent detection. We review detection methods, including noncoherent, differentially coherent, and coherent detection, as well as a hybrid method. We compare modulation methods encoding information in various degrees of freedom (DOF). Polarization-multiplexed quadrature-amplitude modulation maximizes spectral efficiency and power efficiency, by utilizing all four available DOF, the two field quadratures in the two polarizations. Dual-polarization homodyne or heterodyne downconversion are linear processes that can fully recover the received signal field in these four DOF. When downconverted signals are sampled at the Nyquist rate, compensation of transmission impairments can be performed using digital signal processing (DSP). Linear impairments, including chromatic dispersion and polarization-mode dispersion, can be compensated quasi-exactly using finite impulse response filters. Some nonlinear impairments, such as intra-channel four-wave mixing and nonlinear phase noise, can be compensated partially. Carrier phase recovery can be performed using feedforward methods, even when phase-locked loops may fail due to delay constraints. DSP-based compensation enables a receiver to adapt to time-varying impairments, and facilitates use of advanced forward-error-correction codes. We discuss both single- and multi-carrier system implementations. For a given modulation format, using coherent detection, they offer fundamentally the same spectral efficiency and power efficiency, but may differ in practice, because of different impairments and implementation details. With anticipated advances in analog-to-digital converters and integrated circuit technology, DSP-based coherent receivers at bit rates up to 100 Gbit/s should become practical within the next few years.

Coherent optical OFDM: theory and design.

Abstract: Coherent optical OFDM (CO-OFDM) has recently been proposed and the proof-of-concept transmission experiments have shown its extreme robustness against chromatic dispersion and polarization mode dispersion. In this paper, we first review the theoretical fundamentals for CO-OFDM and its channel model in a 2x2 MIMO-OFDM representation. We then present various design choices for CO-OFDM systems and perform the nonlinearity analysis for RF-to-optical up-converter. We also show the receiver-based digital signal processing to mitigate self-phase-modulation (SPM) and Gordon-Mollenauer phase noise, which is equivalent to the midspan phase conjugation.

Coherent Equalization and POLMUX-RZ-DQPSK for Robust 100-GE Transmission

Abstract: We discuss the use of a coherent digital receiver for the compensation of linear transmission impairments and polarization demultiplexing in a transmission system compatible with a future 100-Gb/s Ethernet standard. We present experimental results on the transmission performance of 111 Gbit/s POLMUX-RZ-DQPSK. For a dense WDM setup with channels carrying 111 Gbit/s with a 50 GHz channel spacing (2.0 bits/s/Hz), we show the feasibility of 2375 km transmission. This is enabled through coherent detection which results in excellent noise performance, and subsequent electronic equalization which provides the high tolerance to polarization mode dispersion and chromatic dispersion (CD). Furthermore, we discuss the impact of sampling and digital signal processing with either 1 or 2 samples/bit. We show that when combined with low-pass electrical filtering, 1 sample/bit signal processing is sufficient to obtain a large tolerance towards CD. The proposed modulation and detection techniques enable 111 Gbit/s transmission that is directly compatible with the existing 10 Gbit/s infrastructure.

Electronic post-compensation of WDM transmission impairments using coherent detection and digital signal processing.

Abstract: A universal post-compensation scheme for fiber impairments in wavelength-division multiplexing (WDM) systems is proposed based on coherent detection and digital signal processing (DSP). Transmission of 10 x 10 Gbit/s binary-phase-shift-keying (BPSK) signals at a channel spacing of 20 GHz over 800 km dispersion shifted fiber (DSF) has been demonstrated numerically.

Compressive Sampling and Lossy Compression

Abstract: Recent results in compressive sampling have shown that sparse signals can be recovered from a small number of random measurements. This property raises the question of whether random measurements can provide an efficient representation of sparse signals in an information-theoretic sense. Through both theoretical and experimental results, we show that encoding a sparse signal through simple scalar quantization of random measurements incurs a significant penalty relative to direct or adaptive encoding of the sparse signal. Information theory provides alternative quantization strategies, but they come at the cost of much greater estimation complexity.

FPGA-based Implementation of Signal Processing Systems

Abstract: Field programmable gate arrays (FPGAs) are an increasingly popular technology for implementing digital signal processing (DSP) systems. By allowing designers to create circuit architectures developed for the specific applications, high levels of performance can be achieved for many DSP applications providing considerable improvements over conventional microprocessor and dedicated DSP processor solutions. The book addresses the key issue in this process specifically, the methods and tools needed for the design, optimization and implementation of DSP systems in programmable FPGA hardware. It presents a review of the leading-edge techniques in this field, analyzing advanced DSP-based design flows for both signal flow graph- (SFG-) based and dataflow-based implementation, system on chip (SoC) aspects, and future trends and challenges for FPGAs. The automation of the techniques for component architectural synthesis, computational models, and the reduction of energy consumption to help improve FPGA performance, are given in detail. Written from a system level design perspective and with a DSP focus, the authors present many practical application examples of complex DSP implementation, involving: high-performance computing e.g. matrix operations such as matrix multiplication; high-speed filtering including finite impulse response (FIR) filters and wave digital filters (WDFs); adaptive filtering e.g. recursive least squares (RLS) filtering; transforms such as the fast Fourier transform (FFT). FPGA-based Implementation of Signal Processing Systems is an important reference for practising engineers and researchers working on the design and development of DSP systems for radio, telecommunication, information, audio-visual and security applications. Senior level electrical and computer engineering graduates taking courses in signal processing or digital signal processing shall also find this volume of interest.

Continuous-Time Digital Controller for High-Frequency DC-DC Converters

Abstract: This paper introduces a voltage mode digital controller for low-power high-frequency DC-DC switch-mode power supplies (SMPS) that has fast transient response, approaching physical limitations of a given power stage. In steady state, the controller operates as a conventional pulsewidth modulation regulator and during transients it utilizes a novel fast voltage recovery mechanism, based on real-time processing of the output voltage in digital domain. This continuous-time digital signal processing mechanism is implemented with a very simple processor consisting of a set of asynchronous comparators, delay cells, and combinatorial logic. To eliminate the need for current measurement and calculate the optimal switching sequence of the power stage transistors, the processor performs a capacitor charge balance algorithm, which is based on the detection of the output voltage peak/valley point. The effectiveness of the controller is demonstrated on an experimental 5 W, 5 V to 1.8 V, 400 kHz buck converter. The converter recovers from load transients through a single on-off action of the power switch, virtually reaching the shortest possible time, limited by the values of the power stage filter components only.

Equalization-enhanced phase noise for coherent-detection systems using electronic digital signal processing

Abstract: In coherent optical systems employing electronic digital signal processing, the fiber chromatic dispersion can be gracefully compensated in electronic domain without resorting to optical techniques. Unlike optical dispersion compensator, the electronic equalizer enhances the impairments from the laser phase noise. This equalization-enhanced phase noise (EEPN) imposes a tighter constraint on the receive laser phase noise for transmission systems with high symbol rate and large electronically-compensated chromatic dispersion.

A Continuous-Time ADC/DSP/DAC System With No Clock and With Activity-Dependent Power Dissipation

Abstract: A continuous-time system that converts its analog input to a continuous-time digital representation without sampling, then processes the information digitally without the aid of a clock, is presented. Without sampling there is no aliasing, which reduces the in-band distortion power by not aliasing into band out-of-band distortion components. The 8-bit system, fabricated in a 90 nm CMOS process, utilizes continuous delay elements as part of a programmable transversal FIR filter. The input is encoded by a delta modulator without a clock into a series of non-uniformly spaced tokens, which are processed by the digital continuous-time filter and converted to an analog output using a custom DAC that guarantees there are no glitches in the output waveform. All activity is signal driven, automatically affording dynamic power scaling that tracks input activity.

Electronic Dispersion Compensation

Abstract: The performance of different electronic equalization and processing schemes for 40- and 10-Gb/s optical transmission over single-mode fiber (SMF) are discussed, from the point of their ability to compensate chromatic dispersion (CD) and polarization mode dispersion (PMD). In addition, the impact of fiber nonlinearity and modulation format on equalization is also investigated. The main objective of this paper is to present an overview and a comparison of the performances rather than a detailed explanation of the principles of the different equalization schemes. The equalizers which will be covered are analog equalizer (feedforward and decision feedback type), maximum likelihood sequence estimator (MLSE), electronic precompensation, coherent/intradyne detection with digital signal processing (DSP) equalization, DSP-based optical orthogonal frequency division multiplexing (OFDM), and turbo equalization.

Comparison of Prediction Techniques to Compensate Time Delays Caused by Digital Control of a Three-Phase Buck-Type PWM Rectifier System

Abstract: The digital control of a three-phase three-switch buck-type rectifier system is analyzed in this paper. Three main sources of time delays in the control loop can be identified for the implementation on a digital signal processor (DSP): 1) the delay time due to the sampling of the control quantities; 2) the one due to the calculation time of the DSP; and 3) the one due to the sample-and-hold function of the pulsewidth modulator. Using the buck-type system as an example, the influence of the time delay on the stability of the inner current control loop is discussed, and two prediction methods for time-delay compensation, i.e., a linear prediction and the Smith prediction, are comparatively evaluated. The control performance and the effect of the delay times and the prediction methods are shown by simulation results and through measurements on a 5-kW prototype.

Software-Based Resolver-to-Digital Conversion Using a DSP

Abstract: A simple and cost-effective software-based resolver-to-digital converter using a digital signal processor is presented. The proposed method incorporates software generation of the resolver carrier using a digital filter for synchronous demodulation of the resolver outputs in such a way that there is a substantial savings on hardware like the costly carrier oscillator and associated digital and analog circuits for amplitude demodulators. In addition, because the method does not cause any time delay, the dynamics of the servo control using the scheme are not affected. Furthermore, the method enables the determination of the angle for a complete 360deg shaft rotation with reasonable accuracy using a lookup table that contains entries of only up to 45deg. Computer simulations and experimental results demonstrate the effectiveness and applicability of the proposed scheme.

Signal processing apparatus and signal processing method

Abstract: Disclosed herein is a signal processing apparatus including: a first decimation processing section for generating, based on a digital signal in a first form, a digital signal in a second form; a second decimation processing section for generating, based on the digital signal in the second form, a digital signal in a third form; a first signal processing section for processing the digital signal in the third form; an interpolation processing section for converting a digital signal in the third form outputted from the first signal processing section into a digital signal in the second form; a second signal processing section for processing the digital signal in the second form outputted from the first decimation processing section; and a combining section for combining the digital signals in the second form outputted from the interpolation processing section and the second signal processing section.

CMOS Active Inductors and Transformers: Principle, Implementation, and Applications

Abstract: Many new topologies and circuit design techniques have emerged recently to improve the performance of active inductors, but a comprehensive treatment of the theory, topology, characteristics, and design constraint of CMOS active inductors and transformers, and a detailed examination of their emerging applications in high-speed analog signal processing and data communications over wire and wireless channels, is not available. This book is an attempt to provide an in-depth examination and a systematic presentation of the operation principles and implementation details of CMOS active inductors and transformers, and a detailed examination of their emerging applications in high-speed analog signal processing and data communications over wire and wireless channels. The content of the book is drawn from recently published research papers and are not available in a single, cohesive book. Equal emphasis is given to the theory of CMOS active inductors and transformers, and their emerging applications. Major subjects to be covered in the book include: inductive characteristics in high-speed analog signal processing and data communications, spiral inductors and transformers modeling and limitations, a historical perspective of device synthesis, the topology, characterization, and implementation of CMOS active inductors and transformers, and the application of CMOS active inductors and transformers in high-speed analog and digital signal processing and data communications.

A Novel Architecture of Delta-Sigma Modulator Enabling All-Digital Multiband Multistandard RF Transmitters Design

Abstract: This paper proposes a new architecture of delta-sigma (DS) modulator suitable for RF digital transmitter design. This novel architecture considerably reduces the speed requirements of the digital signal processing block. The novelty lies in the implementation of a specific fully digital up-conversion in combination with a low-pass DS modulator to produce high-frequency digital-like signals, which can be used to drive highly efficient switching-mode power amplifiers. The proposed architecture is suitable for reconfigurable all-digital, multistandard and multiband wireless transmitters. The novel transmitter architecture has been validated using simulation and implemented on a field-programmable gate array development board for two different signals, code division multiple access and orthogonal frequency division multiplex.

Algebraic Signal Processing Theory: 1-D Space

Abstract: In our paper titled ldquoalgebraic signal processing theory: foundation and 1-D Timerdquo appearing in this issue of the IEEE Transactions on Signal Processing, we presented the algebraic signal processing theory, an axiomatic and general framework for linear signal processing. The basic concept in this theory is the signal model defined as the triple (A,M,Phi), where A is a chosen algebra of filters, M an associated A-module of signals, and Phi is a generalization of the z-transform. Each signal model has its own associated set of basic SP concepts, including filtering, spectrum, and Fourier transform. Examples include infinite and finite discrete time where these notions take their well-known forms. In this paper, we use the algebraic theory to develop infinite and finite space signal models. These models are based on a symmetric space shift operator, which is distinct from the standard time shift. We present the space signal processing concepts of filtering or convolution, ldquoz -transform,rdquo spectrum, and Fourier transform. For finite length space signals, we obtain 16 variants of space models, which have the 16 discrete cosine and sine transforms (DCTs/DSTs) as Fourier transforms. Using this novel derivation, we provide missing signal processing concepts associated with the DCTs/DSTs, establish them as precise analogs to the DFT, get deep insight into their origin, and enable the easy derivation of many of their properties including their fast algorithms.

Multirate Filtering for Digital Signal Processing: MATLAB Applications

Abstract: Multirate signal processing techniques are widely used in many areas of modern engineering such as communications, digital audio, measurements, image and signal processing, speech processing, and multimedia. Multirate Filtering for Digital Signal Processing: MATLAB Applications covers basic and advanced approaches in the design and implementation of multirate filtering. This authoritative volume considers the role of filters in multirate systems, provides efficient solutions of finite and infinite impulse response filters for sampling rate conversion, and discusses examples of multirate multilevel filter banks, offering a must-have book for practitioners and scholars in multirate signal processing.

Audio signal processing

Abstract: An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L−R signals.

On Sampling of Band-Limited Signals Associated With the Linear Canonical Transform

Abstract: Sampling is one of the fundamental topics in the signal processing community. Theorems proposed under this topic form the bridge between the continuous-time signals and discrete-time signals. Several sampling theorems, which aid in the reconstruction of signals in the linear canonical transform (LCT) domain, have been proposed in the literature. However, two main practical issues associated with the sampling of the LCT still remain unresolved. The first one relates to the reconstruction of the original signal from nonuniform samples and the other issue relates to the fact that only a finite number of samples are available practically. Focusing on these issues, this paper seeks to address the above from the LCT point of view. First, we extend several previously developed theorems for signals band-limited in the Fourier domain to signals band-limited in the LCT domain, followed by the derivation of the reconstruction formulas for finite uniform or recurrent nonuniform sampling points associated with the LCT. Simulation results and the potential applications of the theorem are also proposed.

Signal Processing Techniques for Knowledge Extraction and Information Fusion

Abstract: This book brings together the latest research achievements from various areas of signal processing and related disciplines in order to consolidate the existing and proposed new directions in DSP based knowledge extraction and information fusion. Within the book contributions presenting both novel algorithms and existing applications, especially those (but not restricted to) on-line processing of real world data are included. The areas of Knowledge Extraction and Information Fusion are naturally linked and aim at detecting and estimating the signal of interest and its parameters, and further at combining measurements from multiple sensors (and associated databases if appropriate) to achieve improved accuracies and more specific inferences which cannot be achieved by using only a single signal modality. The subject therefore is of major interest for modern biomedical, environmental, and industrial applications to provide a state of the art and propose new techniques in order to combine heterogeneous information sources.

Optical coherence tomography imaging system and method

Abstract: An optical imaging system includes an optical radiation source ( 410, 510 ), a frequency clock module outputting frequency clock signals ( 420 ), an optical interferometer ( 430 ), a data acquisition (DAQ) device ( 440 ) triggered by the frequency clock signals, and a computer ( 450 ) to perform multi-dimensional optical imaging of the samples. The frequency clock signals are processed by software or hardware to produce a record containing frequency-time relationship of the optical radiation source ( 410, 510 ) to externally clock the sampling process of the DAQ device ( 440 ). The system may employ over-sampling and various digital signal processing methods to improve image quality. The system further includes multiple stages of routers ( 1418, 1425 ) connecting the light source ( 1410 ) with a plurality of interferometers ( 1420 a- 1420 n) and a DAQ system ( 1450 ) externally clocked by frequency clock signals to perform high-speed multi-channel optical imaging of samples.

Low-delay signal processing based on highly oversampled digital processing

Abstract: A low-delay signal processing system and method are provided which includes a delta-sigma analog-to- digital converter, an oversampling processor, and a delta-sigma digital-to-analog converter. The delta- sigma analog-to-digital converter receives an input or audio signal and generates a digital sample signal at a high oversampling rate. The oversampling processor is connected to the analog-to-digital converter for processing the digital sample signal at the high oversampling rate with low-delay. The delta-sigma digital-to-analog converter is connected to the oversampling processor for receiving the digital sample signal at the high oversampling rate with low-delay for generating an analog signal. The oversampling processor includes a low-delay filter and a programmable delay element. In this manner, the analog signal is produced with a low delay and high accuracy.

Polarization-demultiplexing algorithm in the digital coherent receiver

Abstract: The constant-modulus algorithm has been widely applied to demultiplexing of dual polarizations with a digital coherent receiver. This paper elucidates the physics behind this algorithm, and proposes a modified method of assuring proper polarization demultiplexing.

Power System Frequency Measurement Under Nonstationary Situations

Abstract: A new method for the measurement of the instantaneous power-system frequency is proposed. It is based on the frequency estimation of the voltage signal using three equidistant samples. An algorithm is developed that diminishes the variance of the estimation. The procedure is applied to the case of single- and three-phase networks, and uncertainty in the frequency estimation is obtained with simulated signal and severe conditions of signal quality. A frequency variation has been assumed as plusmn2 Hz around the nominal value, with a maximum rate of change of 1 Hz/s. The uncertainty of 25 mHz and 3.5 mHz has been obtained for single- and three-phase signals, respectively. A low-cost virtual instrument has been developed to make frequency measurements over the actual voltage signal.

A Real-Time Time-Domain EMI Measurement System for Full-Compliance Measurements According to CISPR 16-1-1

Abstract: Time-domain electromagnetic interference (EMI) measurement systems allow measurement time to be reduced by several orders of magnitude. In this paper, a novel real-time operating time-domain EMI measurement system is presented. By the use of several analog-to-digital converters, the dynamic range requested by international electromagnetic compatibility (EMC) standards is achieved. A real-time operating digital signal processing unit is presented. The frequency band that is investigated is subdivided into several subbands. The EMI signal of the complete frequency band is digitized. By a digital down converter, each subband is shifted toward its baseband and low-pass filtered. The low-pass filtered signal is down sampled. The down-sampled signal is processed by a short-time fast Fourier transform. The obtained spectrogram is processed by a parallel implementation of peak, average, and quasi-peak detectors. The dynamic range of the system has been investigated. A comparison of the digital signal processing to the analog signal processing of an EMI receiver is shown. Measurements have been performed in the frequency range 30 MHz to 1 GHz. Such a system can fulfill the international EMC standard CISPR 16-1-1. By the parallel simulation of several thousand EMI receivers, the measurement at several thousand frequency bins can be performed simultaneously. Due to this benefit, the measurement time can be reduced, and further investigations on a device under test can be performed. These investigations are full characterizations, as well as full scans in the final detector mode, which is especially of benefit for highly unstationary emitting devices.

Simultaneous multiple signal reception and transmission using frequency multiplexing and shared processing

Abstract: A novel mechanism for simultaneous multiple signal reception and transmission using frequency multiplexing and shared processing. Multiple RF signals, which may be of various wireless standards, are received using one or more shared processing blocks thereby significantly reducing chip space and power requirements. Shared components include local oscillators, analog to digital converters, digital RX processing and digital baseband processing. In operation, multiple RX front end circuits, one for each desired wireless signal, generate a plurality of IF signals that are frequency multiplexed and combined to create a single combined IF signal. The combined IF signal is processed by a shared processing block. Digital baseband processing is performed on each receive signal to generate respective data outputs. Further, simultaneous full-duplex transmission and reception is performed using a single local oscillator. The phase/frequency modulation of the frequency synthesizer used in the TX is removed from the local oscillator signal for use in the receiver.

Speakers with a digital signal processor

Abstract: A speaker with a digital signal processor is disclosed. In one aspect, a speaker comprises at least one electromechanical transducer configured to convert an electrical audio signal into sound and a digital signal processor configured to process an audio signal and send the processed audio signal to the electromechanical transducer directly or indirectly.